From 4f272341c7a42a71586523f196b242bccde3be8c Mon Sep 17 00:00:00 2001 From: Tobias Hansen Date: Tue, 22 Sep 2009 16:52:08 +0200 Subject: ALSA: snd-usb-us122l: add support for US-144 Adds support for US-144 when attached on USB1.1. Unlike the US-122L it uses both USB interfaces 0 and 1. Signed-off-by: Tobias Hansen Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 75 ++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 67 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index fd44946ce4b..6c7b64a23c1 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -66,6 +66,28 @@ static int us122l_create_usbmidi(struct snd_card *card) iface, &quirk); } +static int us144_create_usbmidi(struct snd_card *card) +{ + static struct snd_usb_midi_endpoint_info quirk_data = { + .out_ep = 4, + .in_ep = 3, + .out_cables = 0x001, + .in_cables = 0x001 + }; + static struct snd_usb_audio_quirk quirk = { + .vendor_name = "US144", + .product_name = NAME_ALLCAPS, + .ifnum = 0, + .type = QUIRK_MIDI_US122L, + .data = &quirk_data + }; + struct usb_device *dev = US122L(card)->chip.dev; + struct usb_interface *iface = usb_ifnum_to_if(dev, 0); + + return snd_usb_create_midi_interface(&US122L(card)->chip, + iface, &quirk); +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -171,6 +193,11 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; + + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->chip.dev, 0); + usb_autopm_get_interface(iface); + } iface = usb_ifnum_to_if(us122l->chip.dev, 1); usb_autopm_get_interface(iface); return 0; @@ -179,8 +206,14 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) { struct us122l *us122l = hw->private_data; - struct usb_interface *iface = usb_ifnum_to_if(us122l->chip.dev, 1); + struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); + + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->chip.dev, 0); + usb_autopm_put_interface(iface); + } + iface = usb_ifnum_to_if(us122l->chip.dev, 1); usb_autopm_put_interface(iface); if (us122l->first == file) us122l->first = NULL; @@ -443,6 +476,13 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->chip.dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + return false; + } + } err = usb_set_interface(us122l->chip.dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -455,7 +495,10 @@ static bool us122l_create_card(struct snd_card *card) if (!us122l_start(us122l, 44100, 256)) return false; - err = us122l_create_usbmidi(card); + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) + err = us144_create_usbmidi(card); + else + err = us122l_create_usbmidi(card); if (err < 0) { snd_printk(KERN_ERR "us122l_create_usbmidi error %i \n", err); us122l_stop(us122l); @@ -542,6 +585,7 @@ static int us122l_usb_probe(struct usb_interface *intf, return err; } + usb_get_intf(usb_ifnum_to_if(device, 0)); usb_get_dev(device); *cardp = card; return 0; @@ -550,9 +594,16 @@ static int us122l_usb_probe(struct usb_interface *intf, static int snd_us122l_probe(struct usb_interface *intf, const struct usb_device_id *id) { + struct usb_device *device = interface_to_usbdev(intf); struct snd_card *card; int err; + if (device->descriptor.idProduct == USB_ID_US144 + && device->speed == USB_SPEED_HIGH) { + snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); + return -ENOENT; + } + snd_printdd(KERN_DEBUG"%p:%i\n", intf, intf->cur_altsetting->desc.bInterfaceNumber); if (intf->cur_altsetting->desc.bInterfaceNumber != 1) @@ -591,7 +642,8 @@ static void snd_us122l_disconnect(struct usb_interface *intf) snd_usbmidi_disconnect(p); } - usb_put_intf(intf); + usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 0)); + usb_put_intf(usb_ifnum_to_if(us122l->chip.dev, 1)); usb_put_dev(us122l->chip.dev); while (atomic_read(&us122l->mmap_count)) @@ -642,6 +694,13 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ + if (us122l->chip.dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->chip.dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + goto unlock; + } + } err = usb_set_interface(us122l->chip.dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); @@ -675,11 +734,11 @@ static struct usb_device_id snd_us122l_usb_id_table[] = { .idVendor = 0x0644, .idProduct = USB_ID_US122L }, -/* { */ /* US-144 maybe works when @USB1.1. Untested. */ -/* .match_flags = USB_DEVICE_ID_MATCH_DEVICE, */ -/* .idVendor = 0x0644, */ -/* .idProduct = USB_ID_US144 */ -/* }, */ + { /* US-144 only works at USB1.1! Disable module ehci-hcd. */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144 + }, { /* terminator */ } }; -- cgit v1.2.3-70-g09d2 From f0968e3f7a8ea30728d2580d3043a30ea9994ec6 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 27 Sep 2009 23:08:40 +0200 Subject: ALSA: sscape: add supoort for SPEA Media FX/Reveal SC-600 Move code from the OSS sscape driver in order to support old Soundscape OEM models. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 6 ++- sound/isa/sscape.c | 116 +++++++++++++++++++++++++++++++++++++---------------- 2 files changed, 86 insertions(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e3777e1..b90fc164a79 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,10 +377,12 @@ config SND_SSCAPE select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 66187122377..b11c35f6aef 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -127,7 +127,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -784,20 +785,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +848,39 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + snd_printd(KERN_INFO "init delay = %d ms\n", d); + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -954,9 +988,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, if (sscape->type == SSCAPE_VIVO) port += 4; - if (dma1 == dma2) - dma2 = -1; - err = snd_wss_create(card, port, -1, irq, dma1, dma2, WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { @@ -1051,21 +1082,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1109,8 +1126,41 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } + + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; + } if (sscape->type != SSCAPE_VIVO) { /* @@ -1141,8 +1191,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,12 +1199,12 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); -- cgit v1.2.3-70-g09d2 From acd47100914b2896d0699febefd077f85c4dd272 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 1 Oct 2009 00:10:34 +0200 Subject: ALSA: sscape: convert to firmware loader framework The conversion solves the problem that firmware size was set to 64KB while non PnP cards have 128KB firmware files. An additional firmware initialization code has been moved from the OSS driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 8 +- include/sound/sscape_ioctl.h | 21 -- sound/isa/Kconfig | 8 +- sound/isa/sscape.c | 328 ++++++++---------------- 4 files changed, 116 insertions(+), 249 deletions(-) delete mode 100644 include/sound/sscape_ioctl.h (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb4518ce..cf985257ae4 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1631,7 +1631,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-sscape ----------------- - Module for ENSONIQ SoundScape PnP cards. + Module for ENSONIQ SoundScape cards. port - Port # (PnP setup) wss_port - WSS Port # (PnP setup) @@ -1640,9 +1640,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) - This module supports multiple cards. ISA PnP must be enabled. - You need sscape_ctl tool in alsa-tools package for loading - the microcode. + This module supports multiple cards. + + The driver requires the firmware loader support on kernel. Module snd-sun-amd7930 (on sparc only) -------------------------------------- diff --git a/include/sound/sscape_ioctl.h b/include/sound/sscape_ioctl.h deleted file mode 100644 index 0d8885969c6..00000000000 --- a/include/sound/sscape_ioctl.h +++ /dev/null @@ -1,21 +0,0 @@ -#ifndef SSCAPE_IOCTL_H -#define SSCAPE_IOCTL_H - - -struct sscape_bootblock -{ - unsigned char code[256]; - unsigned version; -}; - -#define SSCAPE_MICROCODE_SIZE 65536 - -struct sscape_microcode -{ - unsigned char __user *code; -}; - -#define SND_SSCAPE_LOAD_BOOTB _IOWR('P', 100, struct sscape_bootblock) -#define SND_SSCAPE_LOAD_MCODE _IOW ('P', 101, struct sscape_microcode) - -#endif diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b90fc164a79..02fe81ca88f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,9 +372,9 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape and Ensoniq OEM soundcards. @@ -382,7 +382,11 @@ config SND_SSCAPE The PCM audio is supported on SoundScape Classic, Elite, PnP and VIVO cards. The supported OEM cards are SPEA Media FX and Reveal SC-600. - The MIDI support is very experimental. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b11c35f6aef..1ce465cc66a 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,22 +25,26 @@ #include #include #include +#include #include #include #include #include #include -#include #include #include #include -#include - MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); +MODULE_FIRMWARE("sndscape.co0"); +MODULE_FIRMWARE("sndscape.co1"); +MODULE_FIRMWARE("sndscape.co2"); +MODULE_FIRMWARE("sndscape.co3"); +MODULE_FIRMWARE("sndscape.co4"); +MODULE_FIRMWARE("scope.cod"); static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; @@ -142,14 +146,12 @@ struct soundscape { struct resource *wss_res; struct snd_wss *chip; struct snd_mpu401 *mpu; - struct snd_hwdep *hw; /* * The MIDI device won't work until we've loaded * its firmware via a hardware-dependent device IOCTL */ spinlock_t fwlock; - int hw_in_use; unsigned long midi_usage; unsigned char midi_vol; }; @@ -167,12 +169,6 @@ static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) return (struct soundscape *) (mpu->private_data); } -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -393,12 +389,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); - if ((x & 0xfe) == 0xfe) + if (x == 0xfe || x == 0xff) return 1; msleep(10); @@ -420,10 +416,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; @@ -438,14 +434,14 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) * Upload a byte-stream into the SoundScape using DMA channel A. */ static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, + const unsigned char *data, size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -458,7 +454,6 @@ static int upload_dma_data(struct soundscape *s, /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50); sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); @@ -468,35 +463,17 @@ static int upload_dma_data(struct soundscape *s, sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { @@ -512,6 +489,7 @@ static int upload_dma_data(struct soundscape *s, } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) @@ -537,7 +515,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -547,162 +525,69 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} + release_firmware(init_fw); -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR "timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14]; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + snprintf(name, sizeof(name), "sndscape.co%d", version); - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ @@ -920,7 +805,7 @@ static int mpu401_open(struct snd_mpu401 * mpu) spin_lock_irqsave(&sscape->fwlock, flags); - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { + if (sscape->midi_usage == ULONG_MAX) { err = -EBUSY; } else { ++(sscape->midi_usage); @@ -1053,13 +938,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } @@ -1162,29 +1040,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) return -ENXIO; } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; - } - /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) @@ -1222,28 +1077,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card) wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } + + /* + * Enable the master IRQ ... + */ + sscape_write(sscape, GA_INTENA_REG, 0x80); + + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1301,11 +1184,12 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; @@ -1426,12 +1310,12 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); if ((ret = snd_card_register(card)) < 0) { printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; -- cgit v1.2.3-70-g09d2 From bcde1f8a80d1bdfd43fb498996dfa89666fd7fe3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 2 Oct 2009 18:41:29 +0200 Subject: ALSA: sscape: remove MIDI instances counting with limit ULONG_MAX There is no sense to limit open MIDI connections with limit as high as ULONG_MAX. Also, convert more messages to use the snd_printk. Correct few old and misleading comments as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 101 +++++++++++++++-------------------------------------- 1 file changed, 29 insertions(+), 72 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 1ce465cc66a..c739374af20 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -147,12 +147,6 @@ struct soundscape { struct snd_wss *chip; struct snd_mpu401 *mpu; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -164,11 +158,6 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to @@ -179,7 +168,9 @@ static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned lo if (buf) { if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -482,7 +473,8 @@ static int upload_dma_data(struct soundscape *s, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -504,10 +496,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +530,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "Error loading scope.cod"); + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -554,7 +548,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR "timeout reading firmware version\n"); + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -575,12 +570,13 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "Error loading sndscape.co%d", version); + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", init_fw->size >> 10); release_firmware(init_fw); @@ -750,7 +746,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); if ((inb(wss_io) & 0x80) != 0) goto _done; @@ -774,7 +769,6 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! @@ -794,38 +788,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ static int mpu401_open(struct snd_mpu401 * mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->midi_usage == ULONG_MAX) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* @@ -845,8 +814,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; sscape->mpu = mpu; @@ -993,13 +960,13 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } @@ -1036,7 +1003,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1073,8 +1040,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } strcpy(card->driver, "SoundScape"); @@ -1094,7 +1062,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " + snd_printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%lx\n", port[dev]); goto _release_dma; @@ -1191,7 +1159,7 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1250,18 +1218,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * We have found a candidate ISA PnP card. Now we * have to check that it has the devices that we * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. */ - // printk(KERN_INFO "sscape: %s\n", card->name); /* * Check that we still have room for another sound card ... @@ -1272,7 +1229,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1317,7 +1274,7 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, goto _release_card; if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- cgit v1.2.3-70-g09d2 From 1cb0fdebae08f6daaac81197d8dde1746e0a1d96 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 5 Oct 2009 18:18:57 +0200 Subject: ALSA: sscape: force AD1848 codec mode on old Soundscape Old Soundscape cards (pre PnP) work only with AD1848 codecs. If the CS4231 codec is installed it must be used in AD1848 compatible mode. Also, add gameport support and remove an unused mpu field. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/isa/sscape.c | 33 ++++++++++++++++++++++--- 2 files changed, 30 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cf985257ae4..6de56d134ab 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1639,6 +1639,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. mpu_irq - MPU-401 IRQ # (PnP setup) dma - DMA # (PnP setup) dma2 - 2nd DMA # (PnP setup, -1 to disable) + joystick - Enable gameport - 0 = disable (default), 1 = enable This module supports multiple cards. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index c739374af20..279be505b72 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -54,6 +54,7 @@ static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS] __devinitdata; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -79,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -145,7 +149,6 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; unsigned char midi_vol; }; @@ -815,7 +818,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -836,12 +838,30 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; + + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (sscape->type == SSCAPE_VIVO) + case SSCAPE_VIVO: port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -927,6 +947,7 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; + int val; const char *name; /* @@ -1026,6 +1047,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT -- cgit v1.2.3-70-g09d2 From ed76f652d5329d9dff0ea7f3953b1357ed7f8e6e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Oct 2009 18:27:28 +0200 Subject: ALSA: sscape - Remove invalid __devinitdata to module parameters Module parameters shouldn't be marked as __devinitdata since they can be referred via sysfs even after probing. Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be505b72..579a59b9e47 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -46,15 +46,15 @@ MODULE_FIRMWARE("sndscape.co3"); MODULE_FIRMWARE("sndscape.co4"); MODULE_FIRMWARE("scope.cod"); -static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; -static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; -static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; -static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; -static bool joystick[SNDRV_CARDS] __devinitdata; +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); -- cgit v1.2.3-70-g09d2 From 6fcfa3959a5f5ecb7c333f54f401575d94eb8172 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 10 Oct 2009 10:27:58 +0200 Subject: ALSA: sscape: coding style fixes Fix coding style errors in the driver. Also, add missing argument for CMD_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 169 ++++++++++++++++++++++++++--------------------------- 1 file changed, 83 insertions(+), 86 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 579a59b9e47..e2d5d2d3ed9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -109,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -166,10 +166,12 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { snd_printk(KERN_ERR "sscape: Failed to allocate " "%lu bytes for DMA\n", @@ -190,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -206,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -219,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -248,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -292,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -311,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -319,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -329,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -350,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -427,13 +434,13 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char *data, - size_t size) +static int upload_dma_data(struct soundscape *s, const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; @@ -443,18 +450,21 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* * Upload the firmware to the SoundScape @@ -472,7 +482,7 @@ static int upload_dma_data(struct soundscape *s, sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); @@ -489,7 +499,8 @@ static int upload_dma_data(struct soundscape *s, /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -591,7 +602,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -601,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -615,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -637,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -738,7 +752,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type == SSCAPE_VIVO) wss_io += 4; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ @@ -762,7 +776,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if ((inb(wss_io) & 0x80) != 0) s->type = MEDIA_FX; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { @@ -778,7 +792,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -789,7 +803,7 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { snd_printk(KERN_ERR "sscape: MIDI disabled, " @@ -803,18 +817,18 @@ static int mpu401_open(struct snd_mpu401 * mpu) /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; mpu->private_data = sscape; @@ -866,19 +880,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -928,7 +929,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1034,7 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1055,6 +1055,10 @@ static int __devinit create_sscape(int dev, struct snd_card *card) sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1093,11 +1097,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) goto _release_dma; } - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); - /* * Initialize mixer */ @@ -1155,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1183,7 +1183,8 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } @@ -1236,20 +1237,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; - /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - */ - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { @@ -1298,7 +1294,8 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, if (ret < 0) goto _release_card; - if ((ret = snd_card_register(card)) < 0) { + ret = snd_card_register(card); + if (ret < 0) { snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } -- cgit v1.2.3-70-g09d2 From abd134db940ddccaf6a61d88cf0841a62b917ab3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 10 Oct 2009 10:25:39 +0200 Subject: ALSA: wss: convert CS4231 mixer to dB scale Convert CS4231 mixer to dB scale after AD1848 mixer. Also, add missing microphone boost control for the AD1848 and correct wrong bits for loopback volume on the AD1848. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 43 ++++++++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5d2ba1b749a..754a2089c65 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2198,6 +2198,7 @@ EXPORT_SYMBOL(snd_wss_put_double); static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); static struct snd_kcontrol_new snd_ad1848_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, @@ -2224,38 +2225,45 @@ WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, +WSS_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), }; static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), WSS_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -2267,15 +2275,16 @@ WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE("Loopback Capture Volume", 0, - CS4231_LOOPBACK, 2, 63, 1) +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, + db_scale_6bit), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, -- cgit v1.2.3-70-g09d2 From 633c7e92bdd54ba939f2bd3b78c72e1e1a1dd077 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 11 Oct 2009 12:38:49 +0200 Subject: ALSA: wss: reuse CS4231 controls for AD1848 The C4231 control set is a superset of the AD1848 control set so reuse the CS4231 controls definitions for the AD1848. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 79 ++++++++++++++----------------------------------- 1 file changed, 23 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 754a2089c65..2ba18978b41 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2200,49 +2200,12 @@ static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, - 7, 7, 1, 1), -WSS_DOUBLE_TLV("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -WSS_DOUBLE("Aux Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, - 0, 0, 15, 0, db_scale_rec_gain), -{ - .name = "Capture Source", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -}, -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, - db_scale_6bit), -}; - static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, db_scale_6bit), -WSS_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE_TLV("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, - db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Aux Playback Volume", 0, @@ -2253,15 +2216,6 @@ WSS_DOUBLE("Aux Playback Switch", 1, WSS_DOUBLE_TLV("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, db_scale_5bit_12db_max), -WSS_SINGLE("Mono Playback Switch", 0, - CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE_TLV("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1, - db_scale_4bit), -WSS_SINGLE("Mono Output Playback Switch", 0, - CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, - CS4231_MONO_CTRL, 5, 1, 0), WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0, db_scale_rec_gain), { @@ -2277,6 +2231,20 @@ WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, db_scale_6bit), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_SINGLE("Mono Playback Switch", 0, + CS4231_MONO_CTRL, 7, 1, 1), +WSS_SINGLE_TLV("Mono Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), +WSS_SINGLE("Mono Output Playback Switch", 0, + CS4231_MONO_CTRL, 6, 1, 1), +WSS_SINGLE("Mono Output Playback Bypass", 0, + CS4231_MONO_CTRL, 5, 1, 0), }; static struct snd_kcontrol_new snd_opti93x_controls[] = { @@ -2343,22 +2311,21 @@ int snd_wss_mixer(struct snd_wss *chip) if (err < 0) return err; } - else if (chip->hardware & WSS_HW_AD1848_MASK) - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_ad1848_controls[idx], - chip)); - if (err < 0) - return err; - } - else - for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { + else { + int count = ARRAY_SIZE(snd_wss_controls); + + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + + for (idx = 0; idx < count; idx++) { err = snd_ctl_add(card, snd_ctl_new1(&snd_wss_controls[idx], chip)); if (err < 0) return err; } + } return 0; } EXPORT_SYMBOL(snd_wss_mixer); -- cgit v1.2.3-70-g09d2 From 8066e51ae7329220f459470a38387f8533e99141 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 11 Oct 2009 12:48:00 +0200 Subject: ALSA: snd_dma_pointer workaround for chipsets with buggy DMA The chipsets with the isa_dma_bridge_buggy set do not stop DMA during DMA counter reads. The DMA counter is read in two 8-bit read steps on x86 platform. Sometimes, such reads happen during higher byte change so the lower byte is already decremented (rolled over) but the higher byte is not. It introduces an error that position is moved 256 bytes ahead of the true position. Thus, the next DMA position read can return a lower value then the previous read. If the DMA position is decreased (reversed) the ALSA subsystem is tricked into the playback underrun error and resets the playback. It results in a "pop" during a playback. Work around the issue by reading the counter twice and choosing a higher value. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/isadma.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 79f0f16af33..950e19ba91f 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -85,16 +85,24 @@ EXPORT_SYMBOL(snd_dma_disable); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) { unsigned long flags; - unsigned int result; + unsigned int result, result1; flags = claim_dma_lock(); clear_dma_ff(dma); if (!isa_dma_bridge_buggy) disable_dma(dma); result = get_dma_residue(dma); + /* + * HACK - read the counter again and choose higher value in order to + * avoid reading during counter lower byte roll over if the + * isa_dma_bridge_buggy is set. + */ + result1 = get_dma_residue(dma); if (!isa_dma_bridge_buggy) enable_dma(dma); release_dma_lock(flags); + if (unlikely(result < result1)) + result = result1; #ifdef CONFIG_SND_DEBUG if (result > size) snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size); -- cgit v1.2.3-70-g09d2 From 68f139204c1a2b10cc292d9cca036ebdbb6730a8 Mon Sep 17 00:00:00 2001 From: Wu Zhangjin Date: Sat, 10 Oct 2009 23:53:49 +0800 Subject: ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency SND_CS5535AUDIO is available on Loongson(MIPS compatible) family machines, and checked it with ARCH=x86_64, no relative compiling warnings & errors, so, remove the platform dependency directly. Reported-by: rixed@happyleptic.org Acked-by: Andres Salomon Signed-off-by: Wu Zhangjin Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fb5ee3cc396..75c602b5b13 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -259,7 +259,6 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help -- cgit v1.2.3-70-g09d2 From a688e4885c1aa6b88ab5ffa64655bacc01749c9e Mon Sep 17 00:00:00 2001 From: Tobias Hansen Date: Mon, 12 Oct 2009 16:24:15 +0200 Subject: ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd This is the correct error number for telling the USB system that this driver is not for the device. Signed-off-by: Tobias Hansen Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 6c7b64a23c1..b54e8ca360d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -601,7 +601,7 @@ static int snd_us122l_probe(struct usb_interface *intf, if (device->descriptor.idProduct == USB_ID_US144 && device->speed == USB_SPEED_HIGH) { snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); - return -ENOENT; + return -ENODEV; } snd_printdd(KERN_DEBUG"%p:%i\n", -- cgit v1.2.3-70-g09d2 From 84ed1a1942e8c28fb4c23a6235ec48672fc43e49 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Fri, 23 Oct 2009 16:03:08 +0200 Subject: ALSA: Cleanup redundant tests on unsigned The variables are unsigned so the test `>= 0' is always true, the `< 0' test always fails. In these cases the other part of the test catches wrapped values. In dac_audio_write() there does not occur a test for wrapped values, but the test appears redundant. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/sh_dac_audio.c | 3 --- sound/pci/ca0106/ca0106_proc.c | 4 ++-- sound/pci/ctxfi/ctatc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 4 ++-- sound/pci/emu10k1/io.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 7 files changed, 8 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b2ed8757542..4153752507e 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -164,9 +164,6 @@ static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count, int free; int nbytes; - if (count < 0) - return -EINVAL; - if (!count) { dac_audio_sync(); return 0; diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d10ec6..15523e60351 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -405,7 +405,7 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) ) + if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3) snd_ca0106_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index b1b3a644f73..6bfce99b42a 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -240,7 +240,7 @@ static int select_rom(unsigned int pitch) } else if (pitch == 0x02000000) { /* pitch == 2 */ return 3; - } else if (pitch >= 0x0 && pitch <= 0x08000000) { + } else if (pitch <= 0x08000000) { /* 0 <= pitch <= 8 */ return 0; } else { diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 36e08bd2b3c..6b8ae7b5cd5 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f9748aff..baa7cd508cd 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if (reg < 0xa0 && val <= 0xffffffff && channel_id <= 3) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index c1a5aa15af8..5ef7080e14d 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -256,7 +256,7 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ - if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ + if (value > 0x3f) /* 0 to 0x3f are values */ return 1; spin_lock_irqsave(&emu->emu_lock, flags); outl(reg, emu->port + A_IOCFG); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5cd72..35606ae6086 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if (reg > 9 && reg != 15) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } -- cgit v1.2.3-70-g09d2 From b7d5d946e50116f4150542f881ac90ac74c28165 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 24 Oct 2009 17:47:33 +0200 Subject: sound: remove OSS Ensoniq SoundScape driver The OSS driver for Ensoniq SoundScape cards is broken after conversion to mutexes and a new ALSA snd-sscape driver handles all devices handled by the OSS one. The ALSA driver was tested with these cards: Spea V7 MediaFX Ensoniq Soundscape Elite Ensoniq Soundscape VIVO (this card is not handled by the OSS driver) Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/oss/Kconfig | 12 - sound/oss/Makefile | 1 - sound/oss/sscape.c | 1480 ---------------------------------------------------- 3 files changed, 1493 deletions(-) delete mode 100644 sound/oss/sscape.c (limited to 'sound') diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index bcf2a0698d5..135a2b77cc4 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -287,18 +287,6 @@ config SOUND_DMAP Say Y unless you have 16MB or more RAM or a PCI sound card. -config SOUND_SSCAPE - tristate "Ensoniq SoundScape support" - help - Answer Y if you have a sound card based on the Ensoniq SoundScape - chipset. Such cards are being manufactured at least by Ensoniq, Spea - and Reveal (Reveal makes also other cards). - - If you compile the driver into the kernel, you have to add - "sscape=,,,," to the kernel command - line. - - config SOUND_VMIDI tristate "Loopback MIDI device support" help diff --git a/sound/oss/Makefile b/sound/oss/Makefile index e0ae4d4d6a5..567b8a74178 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -13,7 +13,6 @@ obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o -obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_MSS) += ad1848.o obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c deleted file mode 100644 index 30c36d1f35d..00000000000 --- a/sound/oss/sscape.c +++ /dev/null @@ -1,1480 +0,0 @@ -/* - * sound/oss/sscape.c - * - * Low level driver for Ensoniq SoundScape - * - * - * Copyright (C) by Hannu Savolainen 1993-1997 - * - * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) - * Version 2 (June 1991). See the "COPYING" file distributed with this software - * for more info. - * - * - * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) - * Sergey Smitienko : ensoniq p'n'p support - * Christoph Hellwig : adapted to module_init/module_exit - * Bartlomiej Zolnierkiewicz : added __init to attach_sscape() - * Chris Rankin : Specify that this module owns the coprocessor - * Arnaldo C. de Melo : added missing restore_flags in sscape_pnp_upload_file - */ - -#include -#include - -#include "sound_config.h" -#include "sound_firmware.h" - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "coproc.h" - -#include "ad1848.h" -#include "mpu401.h" - -/* - * I/O ports - */ -#define MIDI_DATA 0 -#define MIDI_CTRL 1 -#define HOST_CTRL 2 -#define TX_READY 0x02 -#define RX_READY 0x01 -#define HOST_DATA 3 -#define ODIE_ADDR 4 -#define ODIE_DATA 5 - -/* - * Indirect registers - */ - -#define GA_INTSTAT_REG 0 -#define GA_INTENA_REG 1 -#define GA_DMAA_REG 2 -#define GA_DMAB_REG 3 -#define GA_INTCFG_REG 4 -#define GA_DMACFG_REG 5 -#define GA_CDCFG_REG 6 -#define GA_SMCFGA_REG 7 -#define GA_SMCFGB_REG 8 -#define GA_HMCTL_REG 9 - -/* - * DMA channel identifiers (A and B) - */ - -#define SSCAPE_DMA_A 0 -#define SSCAPE_DMA_B 1 - -#define PORT(name) (devc->base+name) - -/* - * Host commands recognized by the OBP microcode - */ - -#define CMD_GEN_HOST_ACK 0x80 -#define CMD_GEN_MPU_ACK 0x81 -#define CMD_GET_BOARD_TYPE 0x82 -#define CMD_SET_CONTROL 0x88 /* Old firmware only */ -#define CMD_GET_CONTROL 0x89 /* Old firmware only */ -#define CTL_MASTER_VOL 0 -#define CTL_MIC_MODE 2 -#define CTL_SYNTH_VOL 4 -#define CTL_WAVE_VOL 7 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d - -#define CMD_ACK 0x80 - -#define IC_ODIE 1 -#define IC_OPUS 2 - -typedef struct sscape_info -{ - int base, irq, dma; - - int codec, codec_irq; /* required to setup pnp cards*/ - int codec_type; - int ic_type; - char* raw_buf; - unsigned long raw_buf_phys; - int buffsize; /* -------------------------- */ - spinlock_t lock; - int ok; /* Properly detected */ - int failed; - int dma_allocated; - int codec_audiodev; - int opened; - int *osp; - int my_audiodev; -} sscape_info; - -static struct sscape_info adev_info = { - 0 -}; - -static struct sscape_info *devc = &adev_info; -static int sscape_mididev = -1; - -/* Some older cards have assigned interrupt bits differently than new ones */ -static char valid_interrupts_old[] = { - 9, 7, 5, 15 -}; - -static char valid_interrupts_new[] = { - 9, 5, 7, 10 -}; - -static char *valid_interrupts = valid_interrupts_new; - -/* - * See the bottom of the driver. This can be set by spea =0/1. - */ - -#ifdef REVEAL_SPEA -static char old_hardware = 1; -#else -static char old_hardware; -#endif - -static void sleep(unsigned howlong) -{ - current->state = TASK_INTERRUPTIBLE; - schedule_timeout(howlong); -} - -static unsigned char sscape_read(struct sscape_info *devc, int reg) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&devc->lock,flags); - outb(reg, PORT(ODIE_ADDR)); - val = inb(PORT(ODIE_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return val; -} - -static void __sscape_write(int reg, int data) -{ - outb(reg, PORT(ODIE_ADDR)); - outb(data, PORT(ODIE_DATA)); -} - -static void sscape_write(struct sscape_info *devc, int reg, int data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - __sscape_write(reg, data); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static unsigned char sscape_pnp_read_codec(sscape_info* devc, unsigned char reg) -{ - unsigned char res; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - res = inb (devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); - return res; - -} - -static void sscape_pnp_write_codec(sscape_info* devc, unsigned char reg, unsigned char data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - outb( data, devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static void host_open(struct sscape_info *devc) -{ - outb((0x00), PORT(HOST_CTRL)); /* Put the board to the host mode */ -} - -static void host_close(struct sscape_info *devc) -{ - outb((0x03), PORT(HOST_CTRL)); /* Put the board to the MIDI mode */ -} - -static int host_write(struct sscape_info *devc, unsigned char *data, int count) -{ - unsigned long flags; - int i, timeout_val; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Send the command and data bytes - */ - - for (i = 0; i < count; i++) - { - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & TX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - outb(data[i], PORT(HOST_DATA)); - } - spin_unlock_irqrestore(&devc->lock,flags); - return 1; -} - -static int host_read(struct sscape_info *devc) -{ - unsigned long flags; - int timeout_val; - unsigned char data; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Read a byte - */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & RX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return -1; - } - data = inb(PORT(HOST_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return data; -} - -#if 0 /* unused */ -static int host_command1(struct sscape_info *devc, int cmd) -{ - unsigned char buf[10]; - buf[0] = (unsigned char) (cmd & 0xff); - return host_write(devc, buf, 1); -} -#endif /* unused */ - - -static int host_command2(struct sscape_info *devc, int cmd, int parm1) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - - return host_write(devc, buf, 2); -} - -static int host_command3(struct sscape_info *devc, int cmd, int parm1, int parm2) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - buf[2] = (unsigned char) (parm2 & 0xff); - return host_write(devc, buf, 3); -} - -static void set_mt32(struct sscape_info *devc, int value) -{ - host_open(devc); - host_command2(devc, CMD_SET_MT32, value ? 1 : 0); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting MT32 mode failed\n"); */ - } - host_close(devc); -} - -static void set_control(struct sscape_info *devc, int ctrl, int value) -{ - host_open(devc); - host_command3(devc, CMD_SET_CONTROL, ctrl, value); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting control (%d) failed\n", ctrl); */ - } - host_close(devc); -} - -static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode) -{ - unsigned char temp; - - if (dma_chan != SSCAPE_DMA_A) - { - printk(KERN_WARNING "soundscape: Tried to use DMA channel != A. Why?\n"); - return; - } - audio_devs[devc->codec_audiodev]->flags &= ~DMA_AUTOMODE; - DMAbuf_start_dma(devc->codec_audiodev, buf, blk_size, mode); - audio_devs[devc->codec_audiodev]->flags |= DMA_AUTOMODE; - - temp = devc->dma << 4; /* Setup DMA channel select bits */ - if (devc->dma <= 3) - temp |= 0x80; /* 8 bit DMA channel */ - - temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); - temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); -} - -static int verify_mpu(struct sscape_info *devc) -{ - /* - * The SoundScape board could be in three modes (MPU, 8250 and host). - * If the card is not in the MPU mode, enabling the MPU driver will - * cause infinite loop (the driver believes that there is always some - * received data in the buffer. - * - * Detect this by looking if there are more than 10 received MIDI bytes - * (0x00) in the buffer. - */ - - int i; - - for (i = 0; i < 10; i++) - { - if (inb(devc->base + HOST_CTRL) & 0x80) - return 1; - - if (inb(devc->base) != 0x00) - return 1; - } - printk(KERN_WARNING "SoundScape: The device is not in the MPU-401 mode\n"); - return 0; -} - -static int sscape_coproc_open(void *dev_info, int sub_device) -{ - if (sub_device == COPR_MIDI) - { - set_mt32(devc, 0); - if (!verify_mpu(devc)) - return -EIO; - } - return 0; -} - -static void sscape_coproc_close(void *dev_info, int sub_device) -{ - struct sscape_info *devc = dev_info; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - if (devc->dma_allocated) - { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ - devc->dma_allocated = 0; - } - spin_unlock_irqrestore(&devc->lock,flags); - return; -} - -static void sscape_coproc_reset(void *dev_info) -{ -} - -static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, int size, int flag) -{ - unsigned long flags; - unsigned char temp; - volatile int done, timeout_val; - static unsigned char codec_dma_bits; - - if (flag & CPF_FIRST) - { - /* - * First block. Have to allocate DMA and to reset the board - * before continuing. - */ - - spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); - - if (devc->dma_allocated == 0) - devc->dma_allocated = 1; - - spin_unlock_irqrestore(&devc->lock,flags); - - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) & 0x3f); /*Reset */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - sscape_read(devc, GA_HMCTL_REG); /* Delay */ - - /* Take board out of reset */ - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) | 0x80); - } - /* - * Transfer one code block using DMA - */ - if (audio_devs[devc->codec_audiodev]->dmap_out->raw_buf == NULL) - { - printk(KERN_WARNING "soundscape: DMA buffer not available\n"); - return 0; - } - memcpy(audio_devs[devc->codec_audiodev]->dmap_out->raw_buf, block, size); - - spin_lock_irqsave(&devc->lock,flags); - - /******** INTERRUPTS DISABLED NOW ********/ - - do_dma(devc, SSCAPE_DMA_A, - audio_devs[devc->codec_audiodev]->dmap_out->raw_buf_phys, - size, DMA_MODE_WRITE); - - /* - * Wait until transfer completes. - */ - - done = 0; - timeout_val = 30; - while (!done && timeout_val-- > 0) - { - int resid; - - if (HZ / 50) - sleep(HZ / 50); - clear_dma_ff(devc->dma); - if ((resid = get_dma_residue(devc->dma)) == 0) - done = 1; - } - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - return 0; - - if (flag & CPF_LAST) - { - /* - * Take the board out of reset - */ - outb((0x00), PORT(HOST_CTRL)); - outb((0x00), PORT(MIDI_CTRL)); - - temp = sscape_read(devc, GA_HMCTL_REG); - temp |= 0x40; - sscape_write(devc, GA_HMCTL_REG, temp); /* Kickstart the board */ - - /* - * Wait until the ODB wakes up - */ - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - - sleep(1); - x = inb(PORT(HOST_DATA)); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); - done = 1; - } - } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: The OBP didn't respond after code download\n"); - return 0; - } - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - sleep(1); - if (inb(PORT(HOST_DATA)) == 0xfe) /* Host startup acknowledge */ - done = 1; - } - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - return 0; - } - printk(KERN_INFO "SoundScape board initialized OK\n"); - set_control(devc, CTL_MASTER_VOL, 100); - set_control(devc, CTL_SYNTH_VOL, 100); - -#ifdef SSCAPE_DEBUG3 - /* - * Temporary debugging aid. Print contents of the registers after - * downloading the code. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - } - return 1; -} - -static int download_boot_block(void *dev_info, copr_buffer * buf) -{ - if (buf->len <= 0 || buf->len > sizeof(buf->data)) - return -EINVAL; - - if (!sscape_download_boot(devc, buf->data, buf->len, buf->flags)) - { - printk(KERN_ERR "soundscape: Unable to load microcode block to the OBP.\n"); - return -EIO; - } - return 0; -} - -static int sscape_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local) -{ - copr_buffer *buf; - int err; - - switch (cmd) - { - case SNDCTL_COPR_RESET: - sscape_coproc_reset(dev_info); - return 0; - - case SNDCTL_COPR_LOAD: - buf = (copr_buffer *) vmalloc(sizeof(copr_buffer)); - if (buf == NULL) - return -ENOSPC; - if (copy_from_user(buf, arg, sizeof(copr_buffer))) - { - vfree(buf); - return -EFAULT; - } - err = download_boot_block(dev_info, buf); - vfree(buf); - return err; - - default: - return -EINVAL; - } -} - -static coproc_operations sscape_coproc_operations = -{ - "SoundScape M68K", - THIS_MODULE, - sscape_coproc_open, - sscape_coproc_close, - sscape_coproc_ioctl, - sscape_coproc_reset, - &adev_info -}; - -static struct resource *sscape_ports; -static int sscape_is_pnp; - -static void __init attach_sscape(struct address_info *hw_config) -{ -#ifndef SSCAPE_REGS - /* - * Config register values for Spea/V7 Media FX and Ensoniq S-2000. - * These values are card - * dependent. If you have another SoundScape based card, you have to - * find the correct values. Do the following: - * - Compile this driver with SSCAPE_DEBUG1 defined. - * - Shut down and power off your machine. - * - Boot with DOS so that the SSINIT.EXE program is run. - * - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed - * when detecting the SoundScape. - * - Modify the following list to use the values printed during boot. - * Undefine the SSCAPE_DEBUG1 - */ -#define SSCAPE_REGS { \ -/* I0 */ 0x00, \ -/* I1 */ 0xf0, /* Note! Ignored. Set always to 0xf0 */ \ -/* I2 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I3 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I4 */ 0xf5, /* Ignored */ \ -/* I5 */ 0x10, \ -/* I6 */ 0x00, \ -/* I7 */ 0x2e, /* I7 MEM config A. Likely to vary between models */ \ -/* I8 */ 0x00, /* I8 MEM config B. Likely to vary between models */ \ -/* I9 */ 0x40 /* Ignored */ \ - } -#endif - - unsigned long flags; - static unsigned char regs[10] = SSCAPE_REGS; - - int i, irq_bits = 0xff; - - if (old_hardware) - { - valid_interrupts = valid_interrupts_old; - conf_printf("Ensoniq SoundScape (old)", hw_config); - } - else - conf_printf("Ensoniq SoundScape", hw_config); - - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff)) - { - printk(KERN_ERR "Invalid IRQ%d\n", hw_config->irq); - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - if (sscape_is_pnp) - release_region(devc->codec, 2); - return; - } - - if (!sscape_is_pnp) { - - spin_lock_irqsave(&devc->lock,flags); - /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ - /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ - /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ - /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); - /* Don't destroy CD-ROM DMA config bits (0xc0) */ - sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0)); - /* CD-ROM config (WSS codec actually) */ - sscape_write(devc, 6, regs[6]); - sscape_write(devc, 7, regs[7]); - sscape_write(devc, 8, regs[8]); - /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */ - sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08); - spin_unlock_irqrestore(&devc->lock,flags); - } -#ifdef SSCAPE_DEBUG2 - /* - * Temporary debugging aid. Print contents of the registers after - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - if (probe_mpu401(hw_config, sscape_ports)) - hw_config->always_detect = 1; - hw_config->name = "SoundScape"; - - hw_config->irq *= -1; /* Negative value signals IRQ sharing */ - attach_mpu401(hw_config, THIS_MODULE); - hw_config->irq *= -1; /* Restore it */ - - if (hw_config->slots[1] != -1) /* The MPU driver installed itself */ - { - sscape_mididev = hw_config->slots[1]; - midi_devs[hw_config->slots[1]]->coproc = &sscape_coproc_operations; - } - sscape_write(devc, GA_INTENA_REG, 0x80); /* Master IRQ enable */ - devc->ok = 1; - devc->failed = 0; -} - -static int detect_ga(sscape_info * devc) -{ - unsigned char save; - - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); - - /* - * First check that the address register of "ODIE" is - * there and that it has exactly 4 writable bits. - * First 4 bits - */ - - if ((save = inb(PORT(ODIE_ADDR))) & 0xf0) - { - DDB(printk("soundscape: Detect error A\n")); - return 0; - } - outb((0x00), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x00) - { - DDB(printk("soundscape: Detect error B\n")); - return 0; - } - outb((0xff), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x0f) - { - DDB(printk("soundscape: Detect error C\n")); - return 0; - } - outb((save), PORT(ODIE_ADDR)); - - /* - * Now verify that some indirect registers return zero on some bits. - * This may break the driver with some future revisions of "ODIE" but... - */ - - if (sscape_read(devc, 0) & 0x0c) - { - DDB(printk("soundscape: Detect error D (%x)\n", sscape_read(devc, 0))); - return 0; - } - if (sscape_read(devc, 1) & 0x0f) - { - DDB(printk("soundscape: Detect error E\n")); - return 0; - } - if (sscape_read(devc, 5) & 0x0f) - { - DDB(printk("soundscape: Detect error F\n")); - return 0; - } - return 1; -} - -static int sscape_read_host_ctrl(sscape_info* devc) -{ - return host_read(devc); -} - -static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) -{ - host_command2(devc, a, b); -} - -static int sscape_alloc_dma(sscape_info *devc) -{ - char *start_addr, *end_addr; - int dma_pagesize; - int sz, size; - struct page *page; - - if (devc->raw_buf != NULL) return 0; /* Already done */ - dma_pagesize = (devc->dma < 4) ? (64 * 1024) : (128 * 1024); - devc->raw_buf = NULL; - devc->buffsize = 8192*4; - if (devc->buffsize > dma_pagesize) devc->buffsize = dma_pagesize; - start_addr = NULL; - /* - * Now loop until we get a free buffer. Try to get smaller buffer if - * it fails. Don't accept smaller than 8k buffer for performance - * reasons. - */ - while (start_addr == NULL && devc->buffsize > PAGE_SIZE) { - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - devc->buffsize = PAGE_SIZE * (1 << sz); - start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA, sz); - if (start_addr == NULL) devc->buffsize /= 2; - } - - if (start_addr == NULL) { - printk(KERN_ERR "sscape pnp init error: Couldn't allocate DMA buffer\n"); - return 0; - } else { - /* make some checks */ - end_addr = start_addr + devc->buffsize - 1; - /* now check if it fits into the same dma-pagesize */ - - if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1)) - || end_addr >= (char *) (MAX_DMA_ADDRESS)) { - printk(KERN_ERR "sscape pnp: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, devc->buffsize); - return 0; - } - } - devc->raw_buf = start_addr; - devc->raw_buf_phys = virt_to_bus(start_addr); - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - SetPageReserved(page); - return 1; -} - -static void sscape_free_dma(sscape_info *devc) -{ - int sz, size; - unsigned long start_addr, end_addr; - struct page *page; - - if (devc->raw_buf == NULL) return; - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - start_addr = (unsigned long) devc->raw_buf; - end_addr = start_addr + devc->buffsize; - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - ClearPageReserved(page); - - free_pages((unsigned long) devc->raw_buf, sz); - devc->raw_buf = NULL; -} - -/* Intel version !!!!!!!!! */ - -static int sscape_start_dma(int chan, unsigned long physaddr, int count, int dma_mode) -{ - unsigned long flags; - - flags = claim_dma_lock(); - disable_dma(chan); - clear_dma_ff(chan); - set_dma_mode(chan, dma_mode); - set_dma_addr(chan, physaddr); - set_dma_count(chan, count); - enable_dma(chan); - release_dma_lock(flags); - return 0; -} - -static void sscape_pnp_start_dma(sscape_info* devc, int arg ) -{ - int reg; - if (arg == 0) reg = 2; - else reg = 3; - - sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); - sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); -} - -static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) -{ - int reg; - unsigned long i; - unsigned char d; - - if (arg == 0) reg = 2; - else reg = 3; - - sleep ( 1 ); - i = 0; - do { - d = sscape_read(devc, reg) & 1; - if ( d == 1) break; - i++; - } while (i < 500000); - d = sscape_read(devc, reg) & 1; - return d; -} - -static int sscape_pnp_alloc_dma(sscape_info* devc) -{ - /* printk(KERN_INFO "sscape: requesting dma\n"); */ - if (request_dma(devc -> dma, "sscape")) return 0; - /* printk(KERN_INFO "sscape: dma channel allocated\n"); */ - if (!sscape_alloc_dma(devc)) { - free_dma(devc -> dma); - return 0; - }; - return 1; -} - -static void sscape_pnp_free_dma(sscape_info* devc) -{ - sscape_free_dma( devc); - free_dma(devc -> dma ); - /* printk(KERN_INFO "sscape: dma released\n"); */ -} - -static int sscape_pnp_upload_file(sscape_info* devc, char* fn) -{ - int done = 0; - int timeout_val; - char* data,*dt; - int len,l; - unsigned long flags; - - sscape_write( devc, 9, sscape_read(devc, 9 ) & 0x3F ); - sscape_write( devc, 2, (devc -> dma << 4) | 0x80 ); - sscape_write( devc, 3, 0x20 ); - sscape_write( devc, 9, sscape_read( devc, 9 ) | 0x80 ); - - len = mod_firmware_load(fn, &data); - if (len == 0) { - printk(KERN_ERR "sscape: file not found: %s\n", fn); - return 0; - } - dt = data; - spin_lock_irqsave(&devc->lock,flags); - while ( len > 0 ) { - if (len > devc -> buffsize) l = devc->buffsize; - else l = len; - len -= l; - memcpy(devc->raw_buf, dt, l); dt += l; - sscape_start_dma(devc->dma, devc->raw_buf_phys, l, 0x48); - sscape_pnp_start_dma ( devc, 0 ); - if (sscape_pnp_wait_dma ( devc, 0 ) == 0) { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - } - - spin_unlock_irqrestore(&devc->lock,flags); - vfree(data); - - outb(0, devc -> base + 2); - outb(0, devc -> base); - - sscape_write ( devc, 9, sscape_read( devc, 9 ) | 0x40); - - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - timeout_val = 5 * HZ; - done = 0; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - - if ( !done ) printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, (devc -> dma << 4) + 0x80); - return 1; -} - -static void __init sscape_pnp_init_hw(sscape_info* devc) -{ - unsigned char midi_irq = 0, sb_irq = 0; - unsigned i; - static char code_file_name[23] = "/sndscape/sndscape.cox"; - - int sscape_joystic_enable = 0x7f; - int sscape_mic_enable = 0; - int sscape_ext_midi = 0; - - if ( !sscape_pnp_alloc_dma(devc) ) { - printk(KERN_ERR "sscape: faild to allocate dma\n"); - return; - } - - for (i = 0; i < 4; i++) { - if ( devc -> irq == valid_interrupts[i] ) - midi_irq = i; - if ( devc -> codec_irq == valid_interrupts[i] ) - sb_irq = i; - } - - sscape_write( devc, 5, 0x50); - sscape_write( devc, 7, 0x2e); - sscape_write( devc, 8, 0x00); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, ( devc -> dma << 4) | 0x80); - - sscape_write (devc, 4, 0xF0 | (midi_irq<<2) | midi_irq); - - i = 0x10; //sscape_read(devc, 9) & (devc->ic_type == IC_ODIE ? 0xf0 : 0xc0); - if (sscape_joystic_enable) i |= 8; - - sscape_write (devc, 9, i); - sscape_write (devc, 6, 0x80); - sscape_write (devc, 1, 0x80); - - if (devc -> codec_type == 2) { - sscape_pnp_write_codec( devc, 0x0C, 0x50); - sscape_pnp_write_codec( devc, 0x10, sscape_pnp_read_codec( devc, 0x10) & 0x3F); - sscape_pnp_write_codec( devc, 0x11, sscape_pnp_read_codec( devc, 0x11) | 0xC0); - sscape_pnp_write_codec( devc, 29, 0x20); - } - - if (sscape_pnp_upload_file(devc, "/sndscape/scope.cod") == 0 ) { - printk(KERN_ERR "sscape: faild to upload file /sndscape/scope.cod\n"); - sscape_pnp_free_dma(devc); - return; - } - - i = sscape_read_host_ctrl( devc ); - - if ( (i & 0x0F) > 7 ) { - printk(KERN_ERR "sscape: scope.cod faild\n"); - sscape_pnp_free_dma(devc); - return; - } - if ( i & 0x10 ) sscape_write( devc, 7, 0x2F); - code_file_name[21] = (char) ( i & 0x0F) + 0x30; - if (sscape_pnp_upload_file( devc, code_file_name) == 0) { - printk(KERN_ERR "sscape: faild to upload file %s\n", code_file_name); - sscape_pnp_free_dma(devc); - return; - } - - if (devc->ic_type != IC_ODIE) { - sscape_pnp_write_codec( devc, 10, (sscape_pnp_read_codec(devc, 10) & 0x7f) | - ( sscape_mic_enable == 0 ? 0x00 : 0x80) ); - } - sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ - sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ - sscape_write_host_ctrl2( devc, 0x8A, sscape_ext_midi); - - sscape_pnp_write_codec ( devc, 6, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 7, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 2, 0x1F ); //WD_CDXVOLL - sscape_pnp_write_codec ( devc, 3, 0x1F ); //WD_CDXVOLR - - if (devc -> codec_type == 1) { - sscape_pnp_write_codec ( devc, 4, 0x1F ); - sscape_pnp_write_codec ( devc, 5, 0x1F ); - sscape_write_host_ctrl2( devc, 0x88, sscape_mic_enable); - } else { - int t; - sscape_pnp_write_codec ( devc, 0x10, 0x1F << 1); - sscape_pnp_write_codec ( devc, 0x11, 0xC0 | (0x1F << 1)); - - t = sscape_pnp_read_codec( devc, 0x00) & 0xDF; - if ( (sscape_mic_enable == 0)) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x00, t); - t = sscape_pnp_read_codec( devc, 0x01) & 0xDF; - if ( (sscape_mic_enable == 0) ) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x01, t); - sscape_pnp_write_codec ( devc, 0x40 | 29 , 0x20); - outb(0, devc -> codec); - } - if (devc -> ic_type == IC_OPUS ) { - int i = sscape_read( devc, 9 ); - sscape_write( devc, 9, i | 3 ); - sscape_write( devc, 3, 0x40); - - if (request_region(0x228, 1, "sscape setup junk")) { - outb(0, 0x228); - release_region(0x228,1); - } - sscape_write( devc, 3, (devc -> dma << 4) | 0x80); - sscape_write( devc, 9, i ); - } - - host_close ( devc ); - sscape_pnp_free_dma(devc); -} - -static int __init detect_sscape_pnp(sscape_info* devc) -{ - long i, irq_bits = 0xff; - unsigned int d; - - DDB(printk("Entered detect_sscape_pnp(%x)\n", devc->base)); - - if (!request_region(devc->codec, 2, "sscape codec")) { - printk(KERN_ERR "detect_sscape_pnp: port %x is not free\n", devc->codec); - return 0; - } - - if ((inb(devc->base + 2) & 0x78) != 0) - goto fail; - - d = inb ( devc -> base + 4) & 0xF0; - if (d & 0x80) - goto fail; - - if (d == 0) { - devc->codec_type = 1; - devc->ic_type = IC_ODIE; - } else if ( (d & 0x60) != 0) { - devc->codec_type = 2; - devc->ic_type = IC_OPUS; - } else if ( (d & 0x40) != 0) { /* WTF? */ - devc->codec_type = 2; - devc->ic_type = IC_ODIE; - } else - goto fail; - - sscape_is_pnp = 1; - - outb(0xFA, devc -> base+4); - if ((inb( devc -> base+4) & 0x9F) != 0x0A) - goto fail; - outb(0xFE, devc -> base+4); - if ( (inb(devc -> base+4) & 0x9F) != 0x0E) - goto fail; - if ( (inb(devc -> base+5) & 0x9F) != 0x0E) - goto fail; - - if (devc->codec_type == 2) { - if (devc->codec != devc->base + 8) { - printk("soundscape warning: incorrect codec port specified\n"); - goto fail; - } - d = 0x10 | (sscape_read(devc, 9) & 0xCF); - sscape_write(devc, 9, d); - sscape_write(devc, 6, 0x80); - } else { - //todo: check codec is not base + 8 - } - - d = (sscape_read(devc, 9) & 0x3F) | 0xC0; - sscape_write(devc, 9, d); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80) ) break; - - d = inb(devc -> codec); - if (d & 0x80) - goto fail; - if ( inb(devc -> codec + 2) == 0xFF) - goto fail; - - sscape_write(devc, 9, sscape_read(devc, 9) & 0x3F ); - - d = inb(devc -> codec) & 0x80; - if ( d == 0) { - printk(KERN_INFO "soundscape: hardware detected\n"); - valid_interrupts = valid_interrupts_new; - } else { - printk(KERN_INFO "soundscape: board looks like media fx\n"); - valid_interrupts = valid_interrupts_old; - old_hardware = 1; - } - - sscape_write( devc, 9, 0xC0 | (sscape_read(devc, 9) & 0x3F) ); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80)) - break; - - sscape_pnp_init_hw(devc); - - for (i = 0; i < 4; i++) - { - if (devc->codec_irq == valid_interrupts[i]) { - irq_bits = i; - break; - } - } - sscape_write(devc, GA_INTENA_REG, 0x00); - sscape_write(devc, GA_DMACFG_REG, 0x50); - sscape_write(devc, GA_DMAA_REG, 0x70); - sscape_write(devc, GA_DMAB_REG, 0x20); - sscape_write(devc, GA_INTCFG_REG, 0xf0); - sscape_write(devc, GA_CDCFG_REG, 0x89 | (devc->dma << 4) | (irq_bits << 1)); - - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 0) | 0x20); - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 1) | 0x20); - - return 1; -fail: - release_region(devc->codec, 2); - return 0; -} - -static int __init probe_sscape(struct address_info *hw_config) -{ - devc->base = hw_config->io_base; - devc->irq = hw_config->irq; - devc->dma = hw_config->dma; - devc->osp = hw_config->osp; - -#ifdef SSCAPE_DEBUG1 - /* - * Temporary debugging aid. Print contents of the registers before - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (old value)\n", i, sscape_read(devc, i)); - } -#endif - devc->failed = 1; - - sscape_ports = request_region(devc->base, 2, "mpu401"); - if (!sscape_ports) - return 0; - - if (!request_region(devc->base + 2, 6, "SoundScape")) { - release_region(devc->base, 2); - return 0; - } - - if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) - return 1; - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - return 0; - } - - if (old_hardware) /* Check that it's really an old Spea/Reveal card. */ - { - unsigned char tmp; - int cc; - - if (!((tmp = sscape_read(devc, GA_HMCTL_REG)) & 0xc0)) - { - sscape_write(devc, GA_HMCTL_REG, tmp | 0x80); - for (cc = 0; cc < 200000; ++cc) - inb(devc->base + ODIE_ADDR); - } - } - return 1; -} - -static int __init init_ss_ms_sound(struct address_info *hw_config) -{ - int i, irq_bits = 0xff; - int ad_flags = 0; - struct resource *ports; - - if (devc->failed) - { - printk(KERN_ERR "soundscape: Card not detected\n"); - return 0; - } - if (devc->ok == 0) - { - printk(KERN_ERR "soundscape: Invalid initialization order.\n"); - return 0; - } - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (irq_bits == 0xff) { - printk(KERN_ERR "soundscape: Invalid MSS IRQ%d\n", hw_config->irq); - return 0; - } - - if (old_hardware) - ad_flags = 0x12345677; /* Tell that we may have a CS4248 chip (Spea-V7 Media FX) */ - else if (sscape_is_pnp) - ad_flags = 0x87654321; /* Tell that we have a soundscape pnp with 1845 chip */ - - ports = request_region(hw_config->io_base, 4, "ad1848"); - if (!ports) { - printk(KERN_ERR "soundscape: ports busy\n"); - return 0; - } - - if (!ad1848_detect(ports, &ad_flags, hw_config->osp)) { - release_region(hw_config->io_base, 4); - return 0; - } - - if (!sscape_is_pnp) /*pnp is already setup*/ - { - /* - * Setup the DMA polarity. - */ - sscape_write(devc, GA_DMACFG_REG, 0x50); - - /* - * Take the gate-array off of the DMA channel. - */ - sscape_write(devc, GA_DMAB_REG, 0x20); - - /* - * Init the AD1848 (CD-ROM) config reg. - */ - sscape_write(devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) | (irq_bits << 1)); - } - - if (hw_config->irq == devc->irq) - printk(KERN_WARNING "soundscape: Warning! The WSS mode can't share IRQ with MIDI\n"); - - hw_config->slots[0] = ad1848_init( - sscape_is_pnp ? "SoundScape" : "SoundScape PNP", - ports, - hw_config->irq, - hw_config->dma, - hw_config->dma, - 0, - devc->osp, - THIS_MODULE); - - - if (hw_config->slots[0] != -1) /* The AD1848 driver installed itself */ - { - audio_devs[hw_config->slots[0]]->coproc = &sscape_coproc_operations; - devc->codec_audiodev = hw_config->slots[0]; - devc->my_audiodev = hw_config->slots[0]; - - /* Set proper routings here (what are they) */ - AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE); - } - -#ifdef SSCAPE_DEBUG5 - /* - * Temporary debugging aid. Print contents of the registers - * after the AD1848 device has been initialized. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x\n", i, sscape_read(devc, i)); - } -#endif - return 1; -} - -static void __exit unload_sscape(struct address_info *hw_config) -{ - release_region(devc->base + 2, 6); - unload_mpu401(hw_config); - if (sscape_is_pnp) - release_region(devc->codec, 2); -} - -static void __exit unload_ss_ms_sound(struct address_info *hw_config) -{ - ad1848_unload(hw_config->io_base, - hw_config->irq, - devc->dma, - devc->dma, - 0); - sound_unload_audiodev(hw_config->slots[0]); -} - -static struct address_info cfg; -static struct address_info cfg_mpu; - -static int __initdata spea = -1; -static int mss = 0; -static int __initdata dma = -1; -static int __initdata irq = -1; -static int __initdata io = -1; -static int __initdata mpu_irq = -1; -static int __initdata mpu_io = -1; - -module_param(dma, int, 0); -module_param(irq, int, 0); -module_param(io, int, 0); -module_param(spea, int, 0); /* spea=0/1 set the old_hardware */ -module_param(mpu_irq, int, 0); -module_param(mpu_io, int, 0); -module_param(mss, int, 0); - -static int __init init_sscape(void) -{ - printk(KERN_INFO "Soundscape driver Copyright (C) by Hannu Savolainen 1993-1996\n"); - - cfg.irq = irq; - cfg.dma = dma; - cfg.io_base = io; - - cfg_mpu.irq = mpu_irq; - cfg_mpu.io_base = mpu_io; - /* WEH - Try to get right dma channel */ - cfg_mpu.dma = dma; - - devc->codec = cfg.io_base; - devc->codec_irq = cfg.irq; - devc->codec_type = 0; - devc->ic_type = 0; - devc->raw_buf = NULL; - spin_lock_init(&devc->lock); - - if (cfg.dma == -1 || cfg.irq == -1 || cfg.io_base == -1) { - printk(KERN_ERR "DMA, IRQ, and IO port must be specified.\n"); - return -EINVAL; - } - - if (cfg_mpu.irq == -1 && cfg_mpu.io_base != -1) { - printk(KERN_ERR "MPU_IRQ must be specified if MPU_IO is set.\n"); - return -EINVAL; - } - - if(spea != -1) { - old_hardware = spea; - printk(KERN_INFO "Forcing %s hardware support.\n", - spea?"new":"old"); - } - if (probe_sscape(&cfg_mpu) == 0) - return -ENODEV; - - attach_sscape(&cfg_mpu); - - mss = init_ss_ms_sound(&cfg); - - return 0; -} - -static void __exit cleanup_sscape(void) -{ - if (mss) - unload_ss_ms_sound(&cfg); - unload_sscape(&cfg_mpu); -} - -module_init(init_sscape); -module_exit(cleanup_sscape); - -#ifndef MODULE -static int __init setup_sscape(char *str) -{ - /* io, irq, dma, mpu_io, mpu_irq */ - int ints[6]; - - str = get_options(str, ARRAY_SIZE(ints), ints); - - io = ints[1]; - irq = ints[2]; - dma = ints[3]; - mpu_io = ints[4]; - mpu_irq = ints[5]; - - return 1; -} - -__setup("sscape=", setup_sscape); -#endif -MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 3c76b4d69bedde5b9e7e42612a7d2ede4ab7fd8d Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 25 Oct 2009 11:05:19 +0100 Subject: ALSA: es18xx: remove snd_card pointer from snd_es18xx structure The snd_card pointer is redundant and code can be easily changed to work without it. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 75 ++++++++++++++++++++++++++++++++---------------------- 1 file changed, 44 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 8cfbff73a83..160752bc2e8 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -121,7 +121,6 @@ struct snd_es18xx { unsigned int dma1_shift; unsigned int dma2_shift; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_a_substream; struct snd_pcm_substream *capture_a_substream; @@ -755,7 +754,9 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream, static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) { - struct snd_es18xx *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; unsigned char status; if (chip->caps & ES18XX_CONTROL) { @@ -805,12 +806,16 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) int split = 0; if (chip->caps & ES18XX_HWV) { split = snd_es18xx_mixer_read(chip, 0x64) & 0x80; - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_volume->id); } if (!split) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } /* ack interrupt */ snd_es18xx_mixer_write(chip, 0x66, 0x00); @@ -1691,8 +1696,11 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = { .pointer = snd_es18xx_capture_pointer, }; -static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct snd_pcm ** rpcm) +static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) { + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; struct snd_pcm *pcm; char str[16]; int err; @@ -1701,9 +1709,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct *rpcm = NULL; sprintf(str, "ES%x", chip->version); if (chip->caps & ES18XX_PCM2) - err = snd_pcm_new(chip->card, str, device, 2, 1, &pcm); + err = snd_pcm_new(card, str, device, 2, 1, &pcm); else - err = snd_pcm_new(chip->card, str, device, 1, 1, &pcm); + err = snd_pcm_new(card, str, device, 1, 1, &pcm); if (err < 0) return err; @@ -1737,7 +1745,7 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) struct snd_audiodrive *acard = card->private_data; struct snd_es18xx *chip = acard->chip; - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); @@ -1758,18 +1766,21 @@ static int snd_es18xx_resume(struct snd_card *card) /* restore PM register, we won't wake till (not 0x07) i/o activity though */ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM); - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } #endif /* CONFIG_PM */ -static int snd_es18xx_free(struct snd_es18xx *chip) +static int snd_es18xx_free(struct snd_card *card) { + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; + release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_ctrl_port); release_and_free_resource(chip->res_mpu_port); if (chip->irq >= 0) - free_irq(chip->irq, (void *) chip); + free_irq(chip->irq, (void *) card); if (chip->dma1 >= 0) { disable_dma(chip->dma1); free_dma(chip->dma1); @@ -1784,8 +1795,7 @@ static int snd_es18xx_free(struct snd_es18xx *chip) static int snd_es18xx_dev_free(struct snd_device *device) { - struct snd_es18xx *chip = device->device_data; - return snd_es18xx_free(chip); + return snd_es18xx_free(device->card); } static int __devinit snd_es18xx_new_device(struct snd_card *card, @@ -1808,7 +1818,6 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); spin_lock_init(&chip->ctrl_lock); - chip->card = card; chip->port = port; chip->mpu_port = mpu_port; chip->fm_port = fm_port; @@ -1818,53 +1827,55 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, chip->audio2_vol = 0x00; chip->active = 0; - if ((chip->res_port = request_region(port, 16, "ES18xx")) == NULL) { - snd_es18xx_free(chip); + chip->res_port = request_region(port, 16, "ES18xx"); + if (chip->res_port == NULL) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1); return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", (void *) chip)) { - snd_es18xx_free(chip); + if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + (void *) card)) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; if (request_dma(dma1, "ES18xx DMA 1")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2); return -EBUSY; } chip->dma2 = dma2; if (snd_es18xx_probe(chip) < 0) { - snd_es18xx_free(chip); + snd_es18xx_free(card); return -ENODEV; } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es18xx_free(chip); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, NULL, &ops); + if (err < 0) { + snd_es18xx_free(card); return err; } *rchip = chip; return 0; } -static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) +static int __devinit snd_es18xx_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_audiodrive *acard = card->private_data; + struct snd_es18xx *chip = acard->chip; int err; unsigned int idx; - card = chip->card; - strcpy(card->mixername, chip->pcm->name); for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_base_controls); idx++) { @@ -2161,10 +2172,12 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) chip->port, irq[dev], dma1[dev]); - if ((err = snd_es18xx_pcm(chip, 0, NULL)) < 0) + err = snd_es18xx_pcm(card, 0, NULL); + if (err < 0) return err; - if ((err = snd_es18xx_mixer(chip)) < 0) + err = snd_es18xx_mixer(card); + if (err < 0) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { -- cgit v1.2.3-70-g09d2 From b14f5de731ae657d498d18d713c6431bfbeefb4b Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 25 Oct 2009 11:10:01 +0100 Subject: ALSA: es18xx: remove snd_audiodrive structure Remove intermediate snd_audiodrive structure between snd_card structure and snd_es18xx. This reduces size of source code and binary driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 71 +++++++++++++++++++----------------------------------- 1 file changed, 25 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 160752bc2e8..5cf42b4d65f 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -139,10 +139,6 @@ struct snd_es18xx { #ifdef CONFIG_PM unsigned char pm_reg; #endif -}; - -struct snd_audiodrive { - struct snd_es18xx *chip; #ifdef CONFIG_PNP struct pnp_dev *dev; struct pnp_dev *devc; @@ -755,8 +751,7 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream, static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) { struct snd_card *card = dev_id; - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; unsigned char status; if (chip->caps & ES18XX_CONTROL) { @@ -1699,8 +1694,7 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = { static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; struct snd_pcm *pcm; char str[16]; int err; @@ -1742,8 +1736,7 @@ static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, #ifdef CONFIG_PM static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -1760,8 +1753,7 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) static int snd_es18xx_resume(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; /* restore PM register, we won't wake till (not 0x07) i/o activity though */ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM); @@ -1773,8 +1765,7 @@ static int snd_es18xx_resume(struct snd_card *card) static int snd_es18xx_free(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_ctrl_port); @@ -1789,7 +1780,6 @@ static int snd_es18xx_free(struct snd_card *card) disable_dma(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } @@ -1802,19 +1792,14 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, unsigned long port, unsigned long mpu_port, unsigned long fm_port, - int irq, int dma1, int dma2, - struct snd_es18xx ** rchip) + int irq, int dma1, int dma2) { - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; static struct snd_device_ops ops = { .dev_free = snd_es18xx_dev_free, }; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); spin_lock_init(&chip->ctrl_lock); @@ -1865,14 +1850,12 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, snd_es18xx_free(card); return err; } - *rchip = chip; return 0; } static int __devinit snd_es18xx_mixer(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; int err; unsigned int idx; @@ -2074,11 +2057,11 @@ static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) return 0; } -static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnp(int dev, struct snd_es18xx *chip, struct pnp_dev *pdev) { - acard->dev = pdev; - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + chip->dev = pdev; + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; } @@ -2104,26 +2087,26 @@ static struct pnp_card_device_id snd_audiodrive_pnpids[] = { MODULE_DEVICE_TABLE(pnp_card, snd_audiodrive_pnpids); -static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnpc(int dev, struct snd_es18xx *chip, struct pnp_card_link *card, const struct pnp_card_device_id *id) { - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + chip->dev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (chip->dev == NULL) return -EBUSY; - acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devc == NULL) + chip->devc = pnp_request_card_device(card, id->devs[1].id, NULL); + if (chip->devc == NULL) return -EBUSY; /* Control port initialization */ - if (pnp_activate_dev(acard->devc) < 0) { + if (pnp_activate_dev(chip->devc) < 0) { snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } snd_printdd("pnp: port=0x%llx\n", - (unsigned long long)pnp_port_start(acard->devc, 0)); - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + (unsigned long long)pnp_port_start(chip->devc, 0)); + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; @@ -2139,24 +2122,20 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, static int snd_es18xx_card_new(int dev, struct snd_card **cardp) { return snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_audiodrive), cardp); + sizeof(struct snd_es18xx), cardp); } static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; struct snd_opl3 *opl3; int err; - if ((err = snd_es18xx_new_device(card, - port[dev], - mpu_port[dev], - fm_port[dev], - irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) + err = snd_es18xx_new_device(card, + port[dev], mpu_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) return err; - acard->chip = chip; sprintf(card->driver, "ES%x", chip->version); -- cgit v1.2.3-70-g09d2 From bcc2c6b7cb320d10c7fcccd87dce87f4384b4332 Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Sun, 1 Nov 2009 11:13:19 +0100 Subject: ALSA: snd-pcsp: add nopcm mode Currently, if the high-res timers are unavailable, snd-pcsp does not initialize. People who choose it over pcspkr, loose their console beeps in that case and get annoyed. With this patch, the console beeps remain regardless of the high-res timers. Additionally, the "nopcm" modparam is added to forcibly disable the PCM capabilities of the driver. Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/drivers/pcsp/pcsp.c | 32 +++++++++++++++--------- sound/drivers/pcsp/pcsp.h | 2 +- sound/drivers/pcsp/pcsp_mixer.c | 33 +++++++++++++++++++------ 4 files changed, 48 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 6de56d134ab..780c213c600 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1454,6 +1454,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for internal PC-Speaker. + nopcm - Disable PC-Speaker PCM sound. Only beeps remain. nforce_wa - enable NForce chipset workaround. Expect bad sound. This module supports system beeps, some kind of PCM playback and diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index b60cef257b5..f165c77d627 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -26,6 +26,7 @@ MODULE_ALIAS("platform:pcspkr"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +static int nopcm; /* Disable PCM capability of the driver */ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); @@ -33,6 +34,8 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable PC-Speaker sound."); +module_param(nopcm, bool, 0444); +MODULE_PARM_DESC(nopcm, "Disable PC-Speaker PCM sound. Only beeps remain."); struct snd_pcsp pcsp_chip; @@ -43,13 +46,16 @@ static int __devinit snd_pcsp_create(struct snd_card *card) int err; int div, min_div, order; - hrtimer_get_res(CLOCK_MONOTONIC, &tp); - if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { - printk(KERN_ERR "PCSP: Timer resolution is not sufficient " - "(%linS)\n", tp.tv_nsec); - printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " - "enabled.\n"); - return -EIO; + if (!nopcm) { + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + printk(KERN_ERR "PCSP: Timer resolution is not sufficient " + "(%linS)\n", tp.tv_nsec); + printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " + "enabled.\n"); + printk(KERN_ERR "PCSP: Turned into nopcm mode.\n"); + nopcm = 1; + } } if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) @@ -107,12 +113,14 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) snd_card_free(card); return err; } - err = snd_pcsp_new_pcm(&pcsp_chip); - if (err < 0) { - snd_card_free(card); - return err; + if (!nopcm) { + err = snd_pcsp_new_pcm(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } } - err = snd_pcsp_new_mixer(&pcsp_chip); + err = snd_pcsp_new_mixer(&pcsp_chip, nopcm); if (err < 0) { snd_card_free(card); return err; diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index 174dd2ff0f2..1e123077923 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -83,6 +83,6 @@ extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); extern void pcsp_sync_stop(struct snd_pcsp *chip); extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); -extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); +extern int snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm); #endif diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 903bc846763..02e05552632 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -119,24 +119,43 @@ static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol, .put = pcsp_##ctl_type##_put, \ } -static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = { +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = { PCSP_MIXER_CONTROL(enable, "Master Playback Switch"), PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"), +}; + +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = { PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), }; -int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip) +static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip, + struct snd_kcontrol_new *ctls, int num) { - struct snd_card *card = chip->card; int i, err; + struct snd_card *card = chip->card; + for (i = 0; i < num; i++) { + err = snd_ctl_add(card, snd_ctl_new1(ctls + i, chip)); + if (err < 0) + return err; + } + return 0; +} + +int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm) +{ + int err; + struct snd_card *card = chip->card; - for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) { - err = snd_ctl_add(card, - snd_ctl_new1(snd_pcsp_controls + i, - chip)); + if (!nopcm) { + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_pcm, + ARRAY_SIZE(snd_pcsp_controls_pcm)); if (err < 0) return err; } + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_spkr, + ARRAY_SIZE(snd_pcsp_controls_spkr)); + if (err < 0) + return err; strcpy(card->mixername, "PC-Speaker"); -- cgit v1.2.3-70-g09d2 From 9dcaa7b25f2c8f6a0485854cd3641f585a154072 Mon Sep 17 00:00:00 2001 From: Rafael Ignacio Zurita Date: Tue, 3 Nov 2009 17:16:27 -0300 Subject: ALSA: sh: add SuperH DAC audio driver for ALSA V4 This is a port of the sound/oss/sh_dac_audio.c driver. The driver uses an on-chip 8-bit D/A converter, which has a speaker connected to one of its channels, found in several ancient HP machines. For interrupts it uses a high-resolution timer (hrtimer). Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx). Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver would be obsolete soon, and it could be removed. Signed-off-by: Rafael Ignacio Zurita Acked-by: Paul Mundt Signed-off-by: Takashi Iwai --- arch/sh/boards/mach-hp6xx/setup.c | 55 ++++ arch/sh/include/mach-common/mach/hp6xx.h | 4 + include/sound/sh_dac_audio.h | 21 ++ sound/sh/Kconfig | 8 + sound/sh/Makefile | 2 + sound/sh/sh_dac_audio.c | 453 +++++++++++++++++++++++++++++++ 6 files changed, 543 insertions(+) create mode 100644 include/sound/sh_dac_audio.h create mode 100644 sound/sh/sh_dac_audio.c (limited to 'sound') diff --git a/arch/sh/boards/mach-hp6xx/setup.c b/arch/sh/boards/mach-hp6xx/setup.c index 8f305b36358..e6dd5e96321 100644 --- a/arch/sh/boards/mach-hp6xx/setup.c +++ b/arch/sh/boards/mach-hp6xx/setup.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -51,9 +52,63 @@ static struct platform_device jornadakbd_device = { .id = -1, }; +static void dac_audio_start(struct dac_audio_pdata *pdata) +{ + u16 v; + u8 v8; + + /* HP Jornada 680/690 speaker on */ + v = inw(HD64461_GPADR); + v &= ~HD64461_GPADR_SPEAKER; + outw(v, HD64461_GPADR); + + /* HP Palmtop 620lx/660lx speaker on */ + v8 = inb(PKDR); + v8 &= ~PKDR_SPEAKER; + outb(v8, PKDR); + + sh_dac_enable(pdata->channel); +} + +static void dac_audio_stop(struct dac_audio_pdata *pdata) +{ + u16 v; + u8 v8; + + /* HP Jornada 680/690 speaker off */ + v = inw(HD64461_GPADR); + v |= HD64461_GPADR_SPEAKER; + outw(v, HD64461_GPADR); + + /* HP Palmtop 620lx/660lx speaker off */ + v8 = inb(PKDR); + v8 |= PKDR_SPEAKER; + outb(v8, PKDR); + + sh_dac_output(0, pdata->channel); + sh_dac_disable(pdata->channel); +} + +static struct dac_audio_pdata dac_audio_platform_data = { + .buffer_size = 64000, + .channel = 1, + .start = dac_audio_start, + .stop = dac_audio_stop, +}; + +static struct platform_device dac_audio_device = { + .name = "dac_audio", + .id = -1, + .dev = { + .platform_data = &dac_audio_platform_data, + } + +}; + static struct platform_device *hp6xx_devices[] __initdata = { &cf_ide_device, &jornadakbd_device, + &dac_audio_device, }; static void __init hp6xx_init_irq(void) diff --git a/arch/sh/include/mach-common/mach/hp6xx.h b/arch/sh/include/mach-common/mach/hp6xx.h index 0d4165a32dc..bcc301ac12f 100644 --- a/arch/sh/include/mach-common/mach/hp6xx.h +++ b/arch/sh/include/mach-common/mach/hp6xx.h @@ -29,6 +29,9 @@ #define PKDR_LED_GREEN 0x10 +/* HP Palmtop 620lx/660lx speaker on/off */ +#define PKDR_SPEAKER 0x20 + #define SCPDR_TS_SCAN_ENABLE 0x20 #define SCPDR_TS_SCAN_Y 0x02 #define SCPDR_TS_SCAN_X 0x01 @@ -42,6 +45,7 @@ #define ADC_CHANNEL_BACKUP 4 #define ADC_CHANNEL_CHARGE 5 +/* HP Jornada 680/690 speaker on/off */ #define HD64461_GPADR_SPEAKER 0x01 #define HD64461_GPADR_PCMCIA0 (0x02|0x08) diff --git a/include/sound/sh_dac_audio.h b/include/sound/sh_dac_audio.h new file mode 100644 index 00000000000..f5deaf1ddb9 --- /dev/null +++ b/include/sound/sh_dac_audio.h @@ -0,0 +1,21 @@ +/* + * SH_DAC specific configuration, for the dac_audio platform_device + * + * Copyright (C) 2009 Rafael Ignacio Zurita + * + * 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. + */ + +#ifndef __INCLUDE_SH_DAC_AUDIO_H +#define __INCLUDE_SH_DAC_AUDIO_H + +struct dac_audio_pdata { + int buffer_size; + int channel; + void (*start)(struct dac_audio_pdata *pd); + void (*stop)(struct dac_audio_pdata *pd); +}; + +#endif /* __INCLUDE_SH_DAC_AUDIO_H */ diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index aed0f90c391..61139f3c161 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -19,5 +19,13 @@ config SND_AICA help ALSA Sound driver for the SEGA Dreamcast console. +config SND_SH_DAC_AUDIO + tristate "SuperH DAC audio support" + depends on SND + depends on CPU_SH3 && HIGH_RES_TIMERS + select SND_PCM + help + Say Y here to include support for the on-chip DAC. + endif # SND_SUPERH diff --git a/sound/sh/Makefile b/sound/sh/Makefile index 8fdcb6e26f0..7d09b5188cf 100644 --- a/sound/sh/Makefile +++ b/sound/sh/Makefile @@ -3,6 +3,8 @@ # snd-aica-objs := aica.o +snd-sh_dac_audio-objs := sh_dac_audio.o # Toplevel Module Dependency obj-$(CONFIG_SND_AICA) += snd-aica.o +obj-$(CONFIG_SND_SH_DAC_AUDIO) += snd-sh_dac_audio.o diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c new file mode 100644 index 00000000000..76d9ad27d91 --- /dev/null +++ b/sound/sh/sh_dac_audio.c @@ -0,0 +1,453 @@ +/* + * sh_dac_audio.c - SuperH DAC audio driver for ALSA + * + * Copyright (c) 2009 by Rafael Ignacio Zurita + * + * + * Based on sh_dac_audio.c (Copyright (C) 2004, 2005 by Andriy Skulysh) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Rafael Ignacio Zurita "); +MODULE_DESCRIPTION("SuperH DAC audio driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}"); + +/* Module Parameters */ +static int index = SNDRV_DEFAULT_IDX1; +static char *id = SNDRV_DEFAULT_STR1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SuperH DAC audio."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SuperH DAC audio."); + +/* main struct */ +struct snd_sh_dac { + struct snd_card *card; + struct snd_pcm_substream *substream; + struct hrtimer hrtimer; + ktime_t wakeups_per_second; + + int rate; + int empty; + char *data_buffer, *buffer_begin, *buffer_end; + int processed; /* bytes proccesed, to compare with period_size */ + int buffer_size; + struct dac_audio_pdata *pdata; +}; + + +static void dac_audio_start_timer(struct snd_sh_dac *chip) +{ + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); +} + +static void dac_audio_stop_timer(struct snd_sh_dac *chip) +{ + hrtimer_cancel(&chip->hrtimer); +} + +static void dac_audio_reset(struct snd_sh_dac *chip) +{ + dac_audio_stop_timer(chip); + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; +} + +static void dac_audio_set_rate(struct snd_sh_dac *chip) +{ + chip->wakeups_per_second = ktime_set(0, 1000000000 / chip->rate); +} + + +/* PCM INTERFACE */ + +static struct snd_pcm_hardware snd_sh_dac_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX), + .formats = SNDRV_PCM_FMTBIT_U8, + .rates = SNDRV_PCM_RATE_8000, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = (48*1024), + .period_bytes_min = 1, + .period_bytes_max = (48*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_sh_dac_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sh_dac_pcm_hw; + + chip->substream = substream; + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + + chip->pdata->start(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + chip->substream = NULL; + + dac_audio_stop_timer(chip); + chip->pdata->stop(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_sh_dac_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_sh_dac_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + + chip->buffer_size = runtime->buffer_size; + memset(chip->data_buffer, 0, chip->pdata->buffer_size); + + return 0; +} + +static int snd_sh_dac_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dac_audio_start_timer(chip); + break; + case SNDRV_PCM_TRIGGER_STOP: + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + dac_audio_stop_timer(chip); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memcpy_toio(chip->data_buffer + b_pos, src, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memset_io(chip->data_buffer + b_pos, 0, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static +snd_pcm_uframes_t snd_sh_dac_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + int pointer = chip->buffer_begin - chip->data_buffer; + + return pointer; +} + +/* pcm ops */ +static struct snd_pcm_ops snd_sh_dac_pcm_ops = { + .open = snd_sh_dac_pcm_open, + .close = snd_sh_dac_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sh_dac_pcm_hw_params, + .hw_free = snd_sh_dac_pcm_hw_free, + .prepare = snd_sh_dac_pcm_prepare, + .trigger = snd_sh_dac_pcm_trigger, + .pointer = snd_sh_dac_pcm_pointer, + .copy = snd_sh_dac_pcm_copy, + .silence = snd_sh_dac_pcm_silence, + .mmap = snd_pcm_lib_mmap_iomem, +}; + +static int __devinit snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) +{ + int err; + struct snd_pcm *pcm; + + /* device should be always 0 for us */ + err = snd_pcm_new(chip->card, "SH_DAC PCM", device, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SH_DAC PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sh_dac_pcm_ops); + + /* buffer size=48K */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 48 * 1024, + 48 * 1024); + + return 0; +} +/* END OF PCM INTERFACE */ + + +/* driver .remove -- destructor */ +static int snd_sh_dac_remove(struct platform_device *devptr) +{ + snd_card_free(platform_get_drvdata(devptr)); + platform_set_drvdata(devptr, NULL); + + return 0; +} + +/* free -- it has been defined by create */ +static int snd_sh_dac_free(struct snd_sh_dac *chip) +{ + /* release the data */ + kfree(chip->data_buffer); + kfree(chip); + + return 0; +} + +static int snd_sh_dac_dev_free(struct snd_device *device) +{ + struct snd_sh_dac *chip = device->device_data; + + return snd_sh_dac_free(chip); +} + +static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) +{ + struct snd_sh_dac *chip = container_of(handle, struct snd_sh_dac, + hrtimer); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + ssize_t b_ps = frames_to_bytes(runtime, runtime->period_size); + + if (!chip->empty) { + sh_dac_output(*chip->buffer_begin, chip->pdata->channel); + chip->buffer_begin++; + + chip->processed++; + if (chip->processed >= b_ps) { + chip->processed -= b_ps; + snd_pcm_period_elapsed(chip->substream); + } + + if (chip->buffer_begin == (chip->data_buffer + + chip->buffer_size - 1)) + chip->buffer_begin = chip->data_buffer; + + if (chip->buffer_begin == chip->buffer_end) + chip->empty = 1; + + } + + if (!chip->empty) + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); + + return HRTIMER_NORESTART; +} + +/* create -- chip-specific constructor for the cards components */ +static int __devinit snd_sh_dac_create(struct snd_card *card, + struct platform_device *devptr, + struct snd_sh_dac **rchip) +{ + struct snd_sh_dac *chip; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_sh_dac_dev_free, + }; + + *rchip = NULL; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + chip->hrtimer.function = sh_dac_audio_timer; + + dac_audio_reset(chip); + chip->rate = 8000; + dac_audio_set_rate(chip); + + chip->pdata = devptr->dev.platform_data; + + chip->data_buffer = kmalloc(chip->pdata->buffer_size, GFP_KERNEL); + if (chip->data_buffer == NULL) { + kfree(chip); + return -ENOMEM; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sh_dac_free(chip); + return err; + } + + *rchip = chip; + + return 0; +} + +/* driver .probe -- constructor */ +static int __devinit snd_sh_dac_probe(struct platform_device *devptr) +{ + struct snd_sh_dac *chip; + struct snd_card *card; + int err; + + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "cannot allocate the card\n"); + return err; + } + + err = snd_sh_dac_create(card, devptr, &chip); + if (err < 0) + goto probe_error; + + err = snd_sh_dac_pcm(chip, 0); + if (err < 0) + goto probe_error; + + strcpy(card->driver, "snd_sh_dac"); + strcpy(card->shortname, "SuperH DAC audio driver"); + printk(KERN_INFO "%s %s", card->longname, card->shortname); + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + snd_printk("ALSA driver for SuperH DAC audio"); + + platform_set_drvdata(devptr, card); + return 0; + +probe_error: + snd_card_free(card); + return err; +} + +/* + * "driver" definition + */ +static struct platform_driver driver = { + .probe = snd_sh_dac_probe, + .remove = snd_sh_dac_remove, + .driver = { + .name = "dac_audio", + }, +}; + +static int __init sh_dac_init(void) +{ + return platform_driver_register(&driver); +} + +static void __exit sh_dac_exit(void) +{ + platform_driver_unregister(&driver); +} + +module_init(sh_dac_init); +module_exit(sh_dac_exit); -- cgit v1.2.3-70-g09d2 From d355c82a0191d5a3e971bd5af96cc81fe3ed25b9 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 3 Nov 2009 15:47:25 +0100 Subject: ALSA: rename "PC Speaker" and "PC Beep" controls to "Beep" To avoid confusion in control names for the standard analog PC Beep generator using a small Internal PC Speaker, rename all related "PC Speaker" and "PC Beep" controls to "Beep" only. This name is more universal and can be also used on more platforms without confusion. Introduce also "Internal Speaker" in ControlNames.txt for systems with full-featured build-in internal speaker. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ControlNames.txt | 3 ++- sound/core/oss/mixer_oss.c | 3 ++- sound/drivers/pcsp/pcsp_mixer.c | 2 +- sound/isa/cmi8330.c | 4 ++-- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/sb/sb_mixer.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 6 +++--- sound/pci/ac97/ac97_patch.c | 12 ++++++------ sound/pci/azt3328.c | 4 ++-- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- sound/pci/cmipci.c | 4 ++-- sound/pci/emu10k1/emumixer.c | 4 ++-- sound/pci/es1938.c | 2 +- sound/pci/hda/patch_cmedia.c | 4 ++-- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/hda/patch_sigmatel.c | 6 +++--- sound/soc/codecs/wm9713.c | 22 +++++++++++----------- 18 files changed, 47 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 5b18298e949..1bb29814a6f 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -18,8 +18,9 @@ SOURCE: Master Master Mono Hardware Master + Internal Speaker Headphone - PC Speaker + Beep (beep generator) Phone Phone Input Phone Output diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 772423889eb..b935ac9dce8 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1251,7 +1251,8 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_SYNTH, "FM", 0 }, /* fallback */ { SOUND_MIXER_SYNTH, "Music", 0 }, /* fallback */ { SOUND_MIXER_PCM, "PCM", 0 }, - { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, + { SOUND_MIXER_SPEAKER, "Beep", 0 }, + { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */ { SOUND_MIXER_LINE, "Line", 0 }, { SOUND_MIXER_MIC, "Mic", 0 }, { SOUND_MIXER_CD, "CD", 0 }, diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 02e05552632..6f633f4f3b9 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -125,7 +125,7 @@ static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = { }; static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = { - PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), + PCSP_MIXER_CONTROL(pcspkr, "Beep Playback Switch"), }; static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip, diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 02f79d25271..8246aae32ab 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -237,7 +237,7 @@ WSS_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), -WSS_SINGLE("PC Speaker Playback Volume", 0, +WSS_SINGLE("Beep Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), WSS_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), @@ -262,7 +262,7 @@ SB_DOUBLE("SB Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, SB_DOUBLE("SB Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), SB_SINGLE("SB Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), SB_SINGLE("SB Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), -SB_SINGLE("SB PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3), +SB_SINGLE("SB Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), SB_DOUBLE("SB Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), SB_DOUBLE("SB Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), SB_SINGLE("SB Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4c6e14f87f2..c76bb00c9d1 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -982,7 +982,7 @@ ES1688_DOUBLE("CD Playback Volume", 0, ES1688_CD_DEV, ES1688_CD_DEV, 4, 0, 15, 0 ES1688_DOUBLE("FM Playback Volume", 0, ES1688_FM_DEV, ES1688_FM_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Mic Playback Volume", 0, ES1688_MIC_DEV, ES1688_MIC_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Aux Playback Volume", 0, ES1688_AUX_DEV, ES1688_AUX_DEV, 4, 0, 15, 0), -ES1688_SINGLE("PC Speaker Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), +ES1688_SINGLE("Beep Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), ES1688_DOUBLE("Capture Volume", 0, ES1688_RECLEV_DEV, ES1688_RECLEV_DEV, 4, 0, 15, 0), ES1688_SINGLE("Capture Switch", 0, ES1688_REC_DEV, 4, 1, 1), { diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 5cf42b4d65f..e5bf3355d2c 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1313,7 +1313,7 @@ ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0) * The chipset specific mixer controls */ static struct snd_kcontrol_new snd_es18xx_opt_speaker = - ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0); + ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 475220bbcc9..318ff0c823e 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -631,7 +631,7 @@ static struct sbmix_elem snd_sb16_ctl_mic_play_switch = static struct sbmix_elem snd_sb16_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3); + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); static struct sbmix_elem snd_sb16_ctl_capture_vol = SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); static struct sbmix_elem snd_sb16_ctl_play_vol = @@ -689,7 +689,7 @@ static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DT019X_SPKR_DEV, 0, 7); + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); static struct sbmix_elem snd_dt019x_ctl_line_play_vol = SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 78288dbfc17..20cb60afb20 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -603,8 +603,8 @@ AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = { -AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1), -AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) +AC97_SINGLE("Beep Playback Switch", AC97_PC_BEEP, 15, 1, 1), +AC97_SINGLE("Beep Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = @@ -1393,7 +1393,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } } - /* build PC Speaker controls */ + /* build Beep controls */ if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && ((ac97->flags & AC97_HAS_PC_BEEP) || snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) { diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 7337abdbe4e..139cf3b2b9d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -800,12 +800,12 @@ AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0), AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1), -AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1), -AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1), -AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1), -AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1), -AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1), -AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1), +AC97_SINGLE("Beep to Headphone Switch", AC97_AUX, 15, 1, 1), +AC97_SINGLE("Beep to Headphone Volume", AC97_AUX, 12, 7, 1), +AC97_SINGLE("Beep to Master Switch", AC97_AUX, 11, 1, 1), +AC97_SINGLE("Beep to Master Volume", AC97_AUX, 8, 7, 1), +AC97_SINGLE("Beep to Mono Switch", AC97_AUX, 7, 1, 1), +AC97_SINGLE("Beep to Mono Volume", AC97_AUX, 4, 7, 1), AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1), AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1), diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 8451a0169f3..69867ace786 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -830,8 +830,8 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0), AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1), AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1), - AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1), - AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), + AZF3328_MIXER_SWITCH("Beep Playback Switch", IDX_MIXER_PCBEEP, 15, 1), + AZF3328_MIXER_VOL_SPECIAL("Beep Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1), AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1), AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index c8c6f437f5b..8f443a9d61e 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -792,8 +792,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Phone Playback Volume", "Video Playback Switch", "Video Playback Volume", - "PC Speaker Playback Switch", - "PC Speaker Playback Volume", + "Beep Playback Switch", + "Beep Playback Volume", "Mono Output Select", "Capture Source", "Capture Switch", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index ddcd4a9fd7e..a312bae08f5 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2302,7 +2302,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), CMIPCI_SB_SW_MONO("Mic Playback Switch", 0), CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0), - CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + CMIPCI_SB_VOL_MONO("Beep Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15), CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0), CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0), @@ -2310,7 +2310,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7), CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7), CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0), - CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), + CMIPCI_DOUBLE("Beep Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0), }; diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index b0fb6c917c3..05afe06e353 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1818,8 +1818,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Master Playback Switch", "Master Capture Switch", "Master Playback Volume", "Master Capture Volume", "Wave Master Playback Volume", "Master Playback Volume", - "PC Speaker Playback Switch", "PC Speaker Capture Switch", - "PC Speaker Playback Volume", "PC Speaker Capture Volume", + "Beep Playback Switch", "Beep Capture Switch", + "Beep Playback Volume", "Beep Capture Volume", "Phone Playback Switch", "Phone Capture Switch", "Phone Playback Volume", "Phone Capture Volume", "Mic Playback Switch", "Mic Capture Switch", diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 820318ee62c..fb83e1ffa5c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1387,7 +1387,7 @@ ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0, db_scale_line), ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0, db_scale_capture), -ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0), +ES1938_SINGLE("Beep Volume", 0, 0x3c, 0, 7, 0), ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0), ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 780e1a72114..85c81feb10c 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -197,8 +197,8 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08ca660dab..08a5b8a5540 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7334,8 +7334,8 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), /* FIXME: this looks suspicious... - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT), */ { } /* end */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 66c0876bf73..426edfa476a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3221,7 +3221,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, /* check for mute support for the the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "PC Beep Playback Switch", + "Beep Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3230,7 +3230,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, /* check to see if there is volume support for the amp */ if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "PC Beep Playback Volume", + "Beep Playback Volume", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3271,7 +3271,7 @@ static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { static int stac92xx_beep_switch_ctl(struct hda_codec *codec) { return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl, - 0, "PC Beep Playback Switch", 0); + 0, "Beep Playback Switch", 0); } #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index abed37acf78..60e360b1046 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), -SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), +SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), @@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w, /* Left Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), @@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), /* Right Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), @@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]); /* Speaker Mixer */ static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), @@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), /* Mono Mixer */ static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), @@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"), static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ - {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, @@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left HP Mixer", NULL, "Capture Headphone Mux"}, /* right HP mixer */ - {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, @@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Capture Mixer", NULL, "Right Capture Source"}, /* speaker mixer */ - {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Beep Playback Switch", "PCBEEP"}, {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, @@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, /* mono mixer */ - {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Beep Playback Switch", "PCBEEP"}, {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, -- cgit v1.2.3-70-g09d2 From ad1cd745060ae2f24026b3b3d09da3426df6ab36 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 4 Nov 2009 14:30:36 +0100 Subject: ALSA: rename "PC Speaker" controls to "Speaker" To unify control names, rename "PC Speaker" to "Speaker" for PPC ALSA drivers. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ControlNames.txt | 2 +- sound/core/oss/mixer_oss.c | 1 + sound/ppc/awacs.c | 12 ++++++------ sound/ppc/burgundy.c | 8 ++++---- sound/ppc/tumbler.c | 2 +- 5 files changed, 13 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt index 1bb29814a6f..fea65bb6269 100644 --- a/Documentation/sound/alsa/ControlNames.txt +++ b/Documentation/sound/alsa/ControlNames.txt @@ -18,7 +18,7 @@ SOURCE: Master Master Mono Hardware Master - Internal Speaker + Speaker (internal speaker) Headphone Beep (beep generator) Phone diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index b935ac9dce8..54e2eb56e4c 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1253,6 +1253,7 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_PCM, "PCM", 0 }, { SOUND_MIXER_SPEAKER, "Beep", 0 }, { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */ + { SOUND_MIXER_SPEAKER, "Speaker", 0 }, /* fallback */ { SOUND_MIXER_LINE, "Line", 0 }, { SOUND_MIXER_MIC, "Mic", 0 }, { SOUND_MIXER_CD, "CD", 0 }, diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 2cc0eda4f20..2e156467b81 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -479,7 +479,7 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Volume", + .name = "Speaker Playback Volume", .info = snd_pmac_awacs_info_volume_amp, .get = snd_pmac_awacs_get_volume_amp, .put = snd_pmac_awacs_put_volume_amp, @@ -525,7 +525,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = { static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_stereo_info, .get = snd_pmac_awacs_get_switch_amp, .put = snd_pmac_awacs_put_switch_amp, @@ -696,17 +696,17 @@ static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = { - AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), + AWACS_VOLUME("Speaker Playback Volume", 4, 6, 1), }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); /* diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 16ed240e423..0accfe49735 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -505,7 +505,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = { MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), @@ -527,7 +527,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = { MASK_ADDR_BURGUNDY_VOLMIC, 16), BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), @@ -549,11 +549,11 @@ BURGUNDY_SWITCH_B("Master Playback Switch", 0, BURGUNDY_OUTPUT_INTERN | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_INTERN, 0, 0); static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 08e584d1453..789f44f4ac7 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = { }; static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_mono_info, .get = tumbler_get_mute_switch, .put = tumbler_put_mute_switch, -- cgit v1.2.3-70-g09d2 From 31cef7076ed9409a33f19ea372d6dc5fdefe27ae Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:34:16 +0100 Subject: control: remove snd_konctrol_volatile::owner_pid field We do not need to save the ID of the process that locked a control because that information is already available in the owner's file data. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/control.h | 1 - sound/core/control.c | 4 +--- 2 files changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/include/sound/control.h b/include/sound/control.h index ef96f07aa03..3517745d0a2 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -56,7 +56,6 @@ struct snd_kcontrol_new { struct snd_kcontrol_volatile { struct snd_ctl_file *owner; /* locked */ - pid_t owner_pid; unsigned int access; /* access rights */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a8b7fabe645..814d2cf1a34 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -672,7 +672,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner_pid; + info->owner = vd->owner->pid; } else { info->owner = -1; } @@ -827,7 +827,6 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, result = -EBUSY; else { vd->owner = file; - vd->owner_pid = current->pid; result = 0; } } @@ -858,7 +857,6 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, result = -EPERM; else { vd->owner = NULL; - vd->owner_pid = 0; result = 0; } } -- cgit v1.2.3-70-g09d2 From 25d27eded1f4fc728e64f443adc339b5229be5d7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Nov 2009 09:35:44 +0100 Subject: control: use reference-counted pid Instead of storing the PID number, take a reference to the task's pid structure. This protects against duplicates due to PID overflows, and using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is correct as seen from the current namespace. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/control.h | 4 +++- sound/core/control.c | 5 +++-- sound/core/pcm.c | 2 +- sound/core/rawmidi.c | 2 +- 4 files changed, 8 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/control.h b/include/sound/control.h index 3517745d0a2..112374dc0c5 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -86,10 +86,12 @@ struct snd_kctl_event { #define snd_kctl_event(n) list_entry(n, struct snd_kctl_event, list) +struct pid; + struct snd_ctl_file { struct list_head list; /* list of all control files */ struct snd_card *card; - pid_t pid; + struct pid *pid; int prefer_pcm_subdevice; int prefer_rawmidi_subdevice; wait_queue_head_t change_sleep; diff --git a/sound/core/control.c b/sound/core/control.c index 814d2cf1a34..73dc10ac33f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -75,7 +75,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->card = card; ctl->prefer_pcm_subdevice = -1; ctl->prefer_rawmidi_subdevice = -1; - ctl->pid = current->pid; + ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; write_lock_irqsave(&card->ctl_files_rwlock, flags); list_add_tail(&ctl->list, &card->ctl_files); @@ -125,6 +125,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); snd_ctl_empty_read_queue(ctl); + put_pid(ctl->pid); kfree(ctl); module_put(card->module); snd_card_file_remove(card, file); @@ -672,7 +673,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner->pid; + info->owner = pid_vnr(vd->owner->pid); } else { info->owner = -1; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c69c60b2a48..8e2c7833614 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -809,7 +809,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { prefer_subdevice = kctl->prefer_pcm_subdevice; if (prefer_subdevice != -1) break; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14c91f..8a81bdafce6 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -415,7 +415,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) subdevice = -1; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { subdevice = kctl->prefer_rawmidi_subdevice; if (subdevice != -1) break; -- cgit v1.2.3-70-g09d2 From 91d12c485b8949cce6c13ab641147c5bc86ce8b9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:12:26 +0200 Subject: sound: rawmidi: fix opened substreams count The substream_opened field is to count the number of opened substreams, not the number of times that any substreams have been opened. Furthermore, all substreams being opened does not imply that the next open would fail, due to the possibility of O_APPEND. With this wrong check, opening a substream multiple times would succeed only if the device had more unopened substreams. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 4e26563431c..818b1299ed9 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -242,8 +242,6 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, return -ENXIO; if (subdevice >= 0 && subdevice >= s->substream_count) return -ENODEV; - if (s->substream_opened >= s->substream_count) - return -EAGAIN; list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { @@ -280,9 +278,9 @@ static int open_substream(struct snd_rawmidi *rmidi, substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; + rmidi->streams[substream->stream].substream_opened++; } substream->use_count++; - rmidi->streams[substream->stream].substream_opened++; return 0; } @@ -466,7 +464,6 @@ static void close_substream(struct snd_rawmidi *rmidi, struct snd_rawmidi_substream *substream, int cleanup) { - rmidi->streams[substream->stream].substream_opened--; if (--substream->use_count) return; @@ -491,6 +488,7 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + rmidi->streams[substream->stream].substream_opened--; } static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) -- cgit v1.2.3-70-g09d2 From e7373b702f6eab35f315e016a4159860a7a4d686 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:13:30 +0100 Subject: sound: pcm: record a substream's owner process Record the pid of the task that opened a PCM substream. For sound cards with hardware mixing, this allows determining which process is associated with a specific substream's volume control. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 3 +++ sound/core/pcm.c | 4 ++++ 2 files changed, 7 insertions(+) (limited to 'sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index de6d981de5d..c83a4a79f16 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -348,6 +348,8 @@ struct snd_pcm_group { /* keep linked substreams */ int count; }; +struct pid; + struct snd_pcm_substream { struct snd_pcm *pcm; struct snd_pcm_str *pstr; @@ -379,6 +381,7 @@ struct snd_pcm_substream { atomic_t mmap_count; unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); + struct pid *pid; #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_substream oss; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8e2c7833614..6884ae031f6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -435,6 +435,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, return; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); + snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); snd_iprintf(buffer, "trigger_time: %ld.%09ld\n", status.trigger_tstamp.tv_sec, status.trigger_tstamp.tv_nsec); snd_iprintf(buffer, "tstamp : %ld.%09ld\n", @@ -900,6 +901,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->private_data = pcm->private_data; substream->ref_count = 1; substream->f_flags = file->f_flags; + substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; return 0; @@ -921,6 +923,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) kfree(runtime->hw_constraints.rules); kfree(runtime); substream->runtime = NULL; + put_pid(substream->pid); + substream->pid = NULL; substream->pstr->substream_opened--; } -- cgit v1.2.3-70-g09d2 From 7584af10cf46e0f4aa1696f1be79fa0f19a945ba Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Nov 2009 10:14:04 +0100 Subject: sound: rawmidi: record a substream's owner process Record the pid of the task that opened a RawMIDI substream. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/rawmidi.h | 2 ++ sound/core/rawmidi.c | 9 +++++++++ 2 files changed, 11 insertions(+) (limited to 'sound') diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index c23c2658570..2480e7d10dc 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -46,6 +46,7 @@ struct snd_rawmidi; struct snd_rawmidi_substream; struct snd_seq_port_info; +struct pid; struct snd_rawmidi_ops { int (*open) (struct snd_rawmidi_substream * substream); @@ -97,6 +98,7 @@ struct snd_rawmidi_substream { struct snd_rawmidi_str *pstr; char name[32]; struct snd_rawmidi_runtime *runtime; + struct pid *pid; /* hardware layer */ struct snd_rawmidi_ops *ops; }; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 818b1299ed9..2f766123b15 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -278,6 +278,7 @@ static int open_substream(struct snd_rawmidi *rmidi, substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; + substream->pid = get_pid(task_pid(current)); rmidi->streams[substream->stream].substream_opened++; } substream->use_count++; @@ -488,6 +489,8 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + put_pid(substream->pid); + substream->pid = NULL; rmidi->streams[substream->stream].substream_opened--; } @@ -1336,6 +1339,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Mode : %s\n" @@ -1357,6 +1363,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Buffer size : %lu\n" -- cgit v1.2.3-70-g09d2 From e8e63cbf9a339c972eeb5ccf8777c8067bdfd869 Mon Sep 17 00:00:00 2001 From: Josh Triplett Date: Fri, 16 Oct 2009 16:03:49 -0700 Subject: oss: Mark loadhex static in hex2hex.c Nothing outside of hex2hex.c references loadhex. Signed-off-by: Josh Triplett --- sound/oss/hex2hex.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/hex2hex.c b/sound/oss/hex2hex.c index 5460faae98c..041ef5c52bc 100644 --- a/sound/oss/hex2hex.c +++ b/sound/oss/hex2hex.c @@ -12,7 +12,7 @@ #define MAX_SIZE (256*1024) unsigned char buf[MAX_SIZE]; -int loadhex(FILE *inf, unsigned char *buf) +static int loadhex(FILE *inf, unsigned char *buf) { int l=0, c, i; -- cgit v1.2.3-70-g09d2 From 657b1989dacf58e83e7a76bca6d4a91a9f294cf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:40:21 +0100 Subject: ALSA: pcm - Use dma_mmap_coherent() if available Use dma_mmap_coherent() for mmapping the buffers allocated via dma_alloc_coherent() if available. Currently, only ARM has this function, so we do temporarily have an ifdef pcm_native.c. This should be handled better globally in future. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 49 +++++++++++++++++++++++++++++++++---------------- 1 file changed, 33 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf2c89..f067c5b906e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -3094,23 +3095,42 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_data = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_data = { + .open = snd_pcm_mmap_data_open, + .close = snd_pcm_mmap_data_close, +}; + +static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, .fault = snd_pcm_mmap_data_fault, }; +#ifndef ARCH_HAS_DMA_MMAP_COHERENT +/* This should be defined / handled globally! */ +#ifdef CONFIG_ARM +#define ARCH_HAS_DMA_MMAP_COHERENT +#endif +#endif + /* * mmap the DMA buffer on RAM */ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - area->vm_ops = &snd_pcm_vm_ops_data; - area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->mmap_count); +#ifdef ARCH_HAS_DMA_MMAP_COHERENT + if (!substream->ops->page && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return dma_mmap_coherent(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ + /* mmap with fault handler */ + area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } @@ -3118,12 +3138,6 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio = -{ - .open = snd_pcm_mmap_data_open, - .close = snd_pcm_mmap_data_close, -}; - int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3133,8 +3147,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif - area->vm_ops = &snd_pcm_vm_ops_data_mmio; - area->vm_private_data = substream; area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3142,7 +3154,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->mmap_count); return 0; } @@ -3159,6 +3170,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, long size; unsigned long offset; size_t dma_bytes; + int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) @@ -3183,10 +3195,15 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (offset > dma_bytes - size) return -EINVAL; + area->vm_ops = &snd_pcm_vm_ops_data; + area->vm_private_data = substream; if (substream->ops->mmap) - return substream->ops->mmap(substream, area); + err = substream->ops->mmap(substream, area); else - return snd_pcm_default_mmap(substream, area); + err = snd_pcm_default_mmap(substream, area); + if (!err) + atomic_inc(&substream->mmap_count); + return err; } EXPORT_SYMBOL(snd_pcm_mmap_data); -- cgit v1.2.3-70-g09d2 From 9eb4a06788a598573c751af1a7e46639afc89513 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:43:39 +0100 Subject: ALSA: pcm - define snd_pcm_default_page_ops() Add a helper (inline) function as the default page ops. Any hacks wrt the page address conversion will be applied in this function. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f067c5b906e..c906be26c31 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3062,6 +3062,13 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file } #endif /* coherent mmap */ +static inline struct page * +snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) +{ + void *vaddr = substream->runtime->dma_area + ofs; + return virt_to_page(vaddr); +} + /* * fault callback for mmapping a RAM page */ @@ -3072,7 +3079,6 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, struct snd_pcm_runtime *runtime; unsigned long offset; struct page * page; - void *vaddr; size_t dma_bytes; if (substream == NULL) @@ -3082,14 +3088,12 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) return VM_FAULT_SIGBUS; - if (substream->ops->page) { + if (substream->ops->page) page = substream->ops->page(substream, offset); - if (!page) - return VM_FAULT_SIGBUS; - } else { - vaddr = runtime->dma_area + offset; - page = virt_to_page(vaddr); - } + else + page = snd_pcm_default_page_ops(substream, offset); + if (!page) + return VM_FAULT_SIGBUS; get_page(page); vmf->page = page; return 0; -- cgit v1.2.3-70-g09d2 From 66b6cfacfc5aa2fda37b0d40cd54931ca5ef8cd7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 12:50:01 +0100 Subject: ALSA: pcm - fix page conversion on non-coherent MIPS arch The non-coherent MIPS arch doesn't give the correct address by a simple virt_to_page() for pages allocated via dma_alloc_coherent(). Original patch by Wu Zhangjin . [Ralf mentioned: "The origins of this patch go back far further. The oldest patch I could find which is a superset of this was written by Atsushi Nemoto and various incarnations of it have been sumitted to and reject by me a number of times through the years."] A proper check of the buffer allocation type was added to avoid the wrong conversion. Note that this doesn't fix perfectly: the pages should be marked with proper pgprot value. This will be done in a future implementation like the conversion to dma_mmap_coherent(). Acked-by: Ralf Baechle Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c906be26c31..e48c5f61857 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3066,6 +3066,10 @@ static inline struct page * snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) { void *vaddr = substream->runtime->dma_area + ofs; +#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return virt_to_page(CAC_ADDR(vaddr)); +#endif return virt_to_page(vaddr); } -- cgit v1.2.3-70-g09d2 From 6985c8877a711c7c307af05203858cb7c3c89d0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 15:04:24 +0100 Subject: ALSA: pcm - fix page conversion on non-coherent PPC arch The non-cohernet PPC arch doesn't give the correct address by a simple virt_to_page() for pages allocated via dma_alloc_coherent(). This patch adds a hack to fix the conversion similarly like MIPS. Note that this doesn't fix perfectly: the pages should be marked with proper pgprot value. This will be done in a future implementation like the conversion to dma_mmap_coherent(). Acked-by: Benjamin Herrenschmidt Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e48c5f61857..29ab46a12e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3069,6 +3069,16 @@ snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) #if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) return virt_to_page(CAC_ADDR(vaddr)); +#endif +#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { + dma_addr_t addr = substream->runtime->dma_addr + ofs; + addr -= get_dma_offset(substream->dma_buffer.dev.dev); + /* assume dma_handle set via pfn_to_phys() in + * mm/dma-noncoherent.c + */ + return pfn_to_page(addr >> PAGE_SHIFT); + } #endif return virt_to_page(vaddr); } -- cgit v1.2.3-70-g09d2 From d6797322231af98b9bb4afb175dd614cf511e5f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Nov 2009 15:08:54 +0100 Subject: ALSA: Remove old DMA-mmap code from arm/devdma.c The call of dma_mmap_coherent() is done in the PCM core now. Signed-off-by: Takashi Iwai --- sound/arm/Makefile | 2 +- sound/arm/aaci.c | 16 ++++------- sound/arm/devdma.c | 80 ------------------------------------------------------ sound/arm/devdma.h | 3 -- 4 files changed, 6 insertions(+), 95 deletions(-) delete mode 100644 sound/arm/devdma.c delete mode 100644 sound/arm/devdma.h (limited to 'sound') diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 5a549ed6c8a..8c0c851d464 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -3,7 +3,7 @@ # obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o devdma.o +snd-aaci-objs := aaci.o obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o snd-pxa2xx-pcm-objs := pxa2xx-pcm.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 1f0f8213e2d..e59372887f3 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -30,7 +30,6 @@ #include #include "aaci.h" -#include "devdma.h" #define DRIVER_NAME "aaci-pl041" @@ -492,7 +491,7 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) /* * Clear out the DMA and any allocated buffers. */ - devdma_hw_free(NULL, substream); + snd_pcm_lib_free_pages(substream); return 0; } @@ -505,8 +504,8 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aaci_pcm_hw_free(substream); - err = devdma_hw_alloc(NULL, substream, - params_buffer_bytes(params)); + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); if (err < 0) goto out; @@ -551,11 +550,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(runtime, bytes); } -static int aaci_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - return devdma_mmap(NULL, substream, vma); -} - /* * Playback specific ALSA stuff @@ -722,7 +716,6 @@ static struct snd_pcm_ops aaci_playback_ops = { .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, @@ -850,7 +843,6 @@ static struct snd_pcm_ops aaci_capture_ops = { .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; /* @@ -1040,6 +1032,8 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, 0, 64 * 104); } return ret; diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c deleted file mode 100644 index 9d1e6665b54..00000000000 --- a/sound/arm/devdma.c +++ /dev/null @@ -1,80 +0,0 @@ -/* - * linux/sound/arm/devdma.c - * - * Copyright (C) 2003-2004 Russell King, All rights reserved. - * - * 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. - * - * ARM DMA shim for ALSA. - */ -#include -#include - -#include -#include - -#include "devdma.h" - -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - - if (runtime->dma_area == NULL) - return; - - if (buf != &substream->dma_buffer) { - dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr); - kfree(runtime->dma_buffer_p); - } - - snd_pcm_set_runtime_buffer(substream, NULL); -} - -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - int ret = 0; - - if (buf) { - if (buf->bytes >= size) - goto out; - devdma_hw_free(dev, substream); - } - - if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) { - buf = &substream->dma_buffer; - } else { - buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); - if (!buf) - goto nomem; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = dev; - buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL); - buf->bytes = size; - buf->private_data = NULL; - - if (!buf->area) - goto free; - } - snd_pcm_set_runtime_buffer(substream, buf); - ret = 1; - out: - runtime->dma_bytes = size; - return ret; - - free: - kfree(buf); - nomem: - return -ENOMEM; -} - -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); -} diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h deleted file mode 100644 index d025329c8a0..00000000000 --- a/sound/arm/devdma.h +++ /dev/null @@ -1,3 +0,0 @@ -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream); -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size); -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma); -- cgit v1.2.3-70-g09d2