October 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Possibility to build alsa driver without having exact kernel source?
by Christian Steffen 07 Oct '09

07 Oct '09

Hello list, I need a new driver for a atom cpu driven device. It needs the "snd_intel_hda" driver to work as it has intels Poulsbo chipset. Currently I have a compiled module, but sound has a constant clicking noise with this one. Therefore I'd like to try an updated alsa driver, which hopefully fixes a problem on Poulsbo chipsets with stepping D0 or newer. My problem is that I only have a compiled running kernel, but no source from it. All I can see (trough uname -a) is: Linux joggler 2.6.30 #4 SMP Tue Jun 23 21:55:08 BST 2009 i686 GNU/Linux So I know, the kernel is a 2.6.30, arch i686. I tried to compile the driver now just with the standard kernel 2.6.30 source from kernel. org. But when trying to load these modules, I get "Invalid module format". Is there any chance to get a working compiled driver without having more information? Thanks! Christian Steffen P Bitte drucken Sie diese Mail nur, wenn dies notwendig ist. please don't print this e-mail unless you really need to

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Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Aha, sorry - I misunderstood what was being asked to test. Two channel audio is working correctly with the development snapshot. Thanks, Gus On Wed 7/10/09 10:16 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 7 Oct 2009 03:12:49 +0000, > wrote: > > > > Hello Takashi Iwai, > > > > I tested your latest snapshot and it did not detect the 6 channel > audio. > This is so per design. The default auto-parser doesn't enable > 6-channel since otherwise you get problems with the automatic HP / > mic > jack detection. > Does it work except for 6-channel things? > thanks, > Takashi > _______________________________________________ > Alsa-devel mailing list > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] ascenario: Small updates for new API.
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. [PATCH 1/3] ascenario: Rename SND_SCN_INT_QOS to SND_SCN_INT_QOS_POWER As proposed by Jaroslav. Emphasis the power consumption relationship. [PATCH 2/3] ascenario: Fix snd_scenario_get_control_id() declaration to match implementation Just different names in declaration and implementation. I favoured _control_id(). [PATCH 3/3] ascenario: Implement snd_scenario_get_control_id() Replace the four explicit functions with a generic one. I was not able to find another way to get the id from the given numid then getting the full list and iterate through it. If there is something easier let me know. :) regards Stefan Schmidt

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[alsa-devel] [RFC] Add scenario management
by Stefan Schmidt 07 Oct '09

07 Oct '09

Hello. This patchset add support for scenario management, aka use cases, in alsa-lib and example programs in alsa-utils. It allows to adjust the current audio kcontrol settings from a prespective of a changed use case like the switch from listening to music on your phone to an incoming phone call and back. To achieve this it offers control aliasing for playback, capture master and switch as well as the option to post- and prefix a sequence of control changes avoiding pops and other unwanted noise. This is of course not meant to replace gstreamer, PulseAudio or a sound server, but is meant to work in tandem with such audio software. We think alsa-lib is the best place for this as it will be used on small embedded system which only use alsa from the full stack of audio libs in linux. We see potential for desktop usage as well though. There is also the support of QoS and the submission to salsa on the TODO list, but we think it is best to gather soem feedback and get the core merged before drifting away with to much other features. Please let me know what can be improved. Project page is here :- http://www.slimlogic.co.uk/?p=40 regards Stefan Schmidt

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Re: [alsa-devel] sscape: general code improvements
by krzysztof.h1＠poczta.fm 07 Oct '09

07 Oct '09

"Takashi Iwai" <tiwai(a)suse.de> pisze: > At Tue, 6 Oct 2009 19:34:38 +0200, > Krzysztof Helt wrote: > > > > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > > > General code and coding style improvements. > > > > Add a missing argument for SET_XXX_MIDI_VOL command. > > > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Do you really want to change oss/sscape.c that you wanted to remove? > No, it is a mistake. > > > --- > > > > What is a standard name for switch to select internal or external midi? > It is very easy to > > add such a functionality to the driver. > > What do you mean exactly to switch int/ext midi? > Usually the driver provides the interfaces for both, and let user choose. > If they are exclusive, it's checked in the open. > Or, if preferred, a control element can be used to choose the > interface... > The driver interface (MPU-401 UART) is only one and there is a command to switch it to the on-board synthetizer or external midi (joysitck port on the card). Regards, Krzysztof --------------------------------------------------------------- Zobacz jak pracuje sie na wysokosciach. Kliknij >>> http://link.interia.pl/f2384

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Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Hello Takashi Iwai, I tested your latest snapshot and it did not detect the 6 channel audio. I have attached the output of alsa-info for analysis. The original patch that I submitted did not detect the coaxial/optical SPDIF interface on this board. I've been running with the ALC662_3ST_6ch_DIG method instead of ALC662_3ST_6ch which was in the original patch. The _DIG method now shows my spdif interface correctly and my 6 channel analog audio is also working correctly. I do not have a receiver to test the coaxial/optical output, though I suspect it works correctly. I've attached a new patch which uses the _DIG method as I believe its the more proper method to use for this hardware layout. Thanks, Gus Bourg On Mon 28/09/09 9:30 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > Could you check whether the very latest alsa-driver snapshot still > doesn't work without your change? > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna > pshot.tar.gzIf it doesn't work, could you give alsa-info.sh output (run > --no-upload option)? > If your fix is mandatory, I'm going to apply it happily. But, then > please give your sign-off, too. > thanks, > Takashi > > Message sent via Atmail Open - http://atmail.org/

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[alsa-devel] alsa-utils cross compilation error
by Harish Kumar 07 Oct '09

07 Oct '09

Hi, I am trying to cross compile the alsa utils for PPC and as well as arm architecture. the compilation of alsa-libs was successful with their respective toolchains. But configuring the alsa-util is failing. I am using alsa-utils-1.0.11rc2,alsa-lib-1.0.11rc2. I am getting the error checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. I feel i have to provide the path of the include files but not sure which path to be passed and how it needs to be passed to configure. Request your help to resolve the same. I invoked the configure using the following command and the output of the configure is as below. root@localhost alsa-utils-1.0.11rc2]# CC=arm-linux-gcc ./configure --target=arm-linux --host=i686-linux configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... gawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gcc accepts -g... yes checking for arm-linux-gcc option to accept ANSI C... none needed checking dependency style of arm-linux-gcc... gcc3 checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld checking if the linker (/opt/GARZ/CROSSTOOLS/arm-2007q3/arm-none-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking whether NLS is requested... yes checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gcc accepts -g... (cached) yes checking for arm-linux-gcc option to accept ANSI C... (cached) none needed checking dependency style of arm-linux-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... checking for ALSA LDFLAGS... -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.9... not present. configure: error: Sufficiently new version of libasound not found. [2]+ Done gedit configure [root@localhost alsa-utils-1.0.11rc2]# Thanks and Regards Harish Kumar.B

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[alsa-devel] ice1724 SPDIF buffer size?
by Robert Hancock 07 Oct '09

07 Oct '09

I noticed that in the ice1724 driver, we set a default and maximum preallocated buffer size of 256K for the "professional" stream, but only 64K for the SPDIF and independent stereo pair streams, even though the hardware supports 256K for all streams. Is there any reason why we can't use 256K for all of them? I made this change here and it seems to work fine (for SPDIF at least, haven't tested stereo). From a power and CPU usage perspective, it makes sense to allow as big a buffer as possible for when latency isn't critical..

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[alsa-devel] sscape: general code improvements
by Krzysztof Helt 07 Oct '09

07 Oct '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> General code and coding style improvements. Add a missing argument for SET_XXX_MIDI_VOL command. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- What is a standard name for switch to select internal or external midi? It is very easy to add such a functionality to the driver. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 279be50..4380e85 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)); } @@ -331,7 +334,8 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) */ static inline 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); } @@ -352,22 +356,25 @@ static void soundscape_free(struct snd_card *c) */ static inline 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; } /* @@ -429,11 +436,12 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) */ static int upload_dma_data(struct soundscape *s, const unsigned char *data, - size_t size) + 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 +451,21 @@ 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 +483,7 @@ 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 +500,8 @@ 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 +603,7 @@ 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 +613,7 @@ 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 +627,18 @@ 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 +651,16 @@ 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 +753,7 @@ 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 +777,7 @@ 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 +793,7 @@ 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 +804,7 @@ 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 +818,18 @@ 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 +881,6 @@ 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 +930,7 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, sscape->chip = chip; } - _error: +_error: return err; } @@ -1155,7 +1157,8 @@ 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 +1186,8 @@ 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 +1240,15 @@ 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 +1297,8 @@ 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; } diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c index 30c36d1..7e06424 100644 --- a/sound/oss/sscape.c +++ b/sound/oss/sscape.c @@ -160,19 +160,27 @@ static void sleep(unsigned howlong) schedule_timeout(howlong); } +static unsigned char __sscape_read(struct sscape_info *devc, int reg) +{ + unsigned char val; + + outb(reg, PORT(ODIE_ADDR)); + val = inb(PORT(ODIE_DATA)); + return val; +} + 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)); + val = __sscape_read(devc, reg); spin_unlock_irqrestore(&devc->lock,flags); return val; } -static void __sscape_write(int reg, int data) +static void __sscape_write(struct sscape_info *devc, int reg, int data) { outb(reg, PORT(ODIE_ADDR)); outb(data, PORT(ODIE_DATA)); @@ -183,7 +191,7 @@ 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); + __sscape_write(devc, reg, data); spin_unlock_irqrestore(&devc->lock,flags); } @@ -342,9 +350,9 @@ static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, in temp |= 0x80; /* 8 bit DMA channel */ temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); + __sscape_write(devc, GA_DMAA_REG, temp); } static int verify_mpu(struct sscape_info *devc) @@ -392,7 +400,7 @@ static void sscape_coproc_close(void *dev_info, int sub_device) spin_lock_irqsave(&devc->lock,flags); if (devc->dma_allocated) { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ + __sscape_write(devc, GA_DMAA_REG, 0x20); /* DMA channel disabled */ devc->dma_allocated = 0; } spin_unlock_irqrestore(&devc->lock,flags); @@ -418,7 +426,7 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, */ spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); + codec_dma_bits = __sscape_read(devc, GA_CDCFG_REG); if (devc->dma_allocated == 0) devc->dma_allocated = 1; @@ -500,11 +508,11 @@ static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, x = inb(PORT(HOST_DATA)); if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); + (printk("Soundscape: Acknowledge = %x\n", x)); done = 1; } } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); + __sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); spin_unlock_irqrestore(&devc->lock,flags); if (!done) @@ -604,6 +612,16 @@ static coproc_operations sscape_coproc_operations = static struct resource *sscape_ports; static int sscape_is_pnp; +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 void __init attach_sscape(struct address_info *hw_config) { #ifndef SSCAPE_REGS @@ -669,22 +687,25 @@ static void __init attach_sscape(struct address_info *hw_config) spin_lock_irqsave(&devc->lock,flags); /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ + __sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 2, 0x20); /* DMA channel disabled */ /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ + __sscape_write(devc, 3, 0x20); /* DMA channel disabled */ /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); + __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)); + __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]); + __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); + __sscape_write(devc, 9, (__sscape_read(devc, 9) & 0xf0) | 0x08); spin_unlock_irqrestore(&devc->lock,flags); + sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ + sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ + sscape_write_host_ctrl2( devc, 0x8A, 0); } #ifdef SSCAPE_DEBUG2 /* @@ -721,7 +742,7 @@ static int detect_ga(sscape_info * devc) { unsigned char save; - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); + printk(KERN_DEBUG "Entered Soundscape detect_ga(%x)\n", devc->base); /* * First check that the address register of "ODIE" is @@ -771,16 +792,6 @@ static int detect_ga(sscape_info * devc) 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; @@ -869,8 +880,8 @@ static void sscape_pnp_start_dma(sscape_info* devc, int arg ) 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); + __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 ) @@ -885,11 +896,11 @@ static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) sleep ( 1 ); i = 0; do { - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; if ( d == 1) break; i++; } while (i < 500000); - d = sscape_read(devc, reg) & 1; + d = __sscape_read(devc, reg) & 1; return d; } @@ -1242,8 +1253,11 @@ static int __init probe_sscape(struct address_info *hw_config) } if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) + printk(KERN_DEBUG "GA detected!\n"); + if (detect_sscape_pnp(devc)) { + printk(KERN_DEBUG "PnP detected!\n"); return 1; + } release_region(devc->base, 2); release_region(devc->base + 2, 6); return 0; ---------------------------------------------------------------------- Zobacz najwiekszy samolot na swiecie! 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[alsa-devel] PCM device volume
by Milan Křápek 07 Oct '09

07 Oct '09

Hi, I am new with alsa. I am developing application that will have audio output on linux systems via alsa. I found many hints when I was trying to play some audio. I used snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); snd_pcm_hw_params_any(pcm_handle, hwparams); to open and set my PCM device for writting audio. But I have got lost when I tried to change my audio volume. I suppose that I need to get some mixer int snd_mixer_open (snd_mixer_t **mixer, int mode) but I could not find how to code that the new mixer will control the PCM device. Pleas is there any tutorial how to do it? Or can somebody show me any code snippet? I have tried to find any similar topic, but without success. Thanks for any advice Milan

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