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[alsa-devel] Problem in dummy driver
by Himanshu Chauhan 21 Mar '07

21 Mar '07

Dear all, I am working on the dummy driver provided with the ALSA. I took it from the linux kernel 2.6.20.1 I build the module and load it. The XMMS seems to play ok (doesn't hang and all) but none of the recording application seem to record from the driver. Can the driver in current state work as the loop back cable between applications? Thanks in advance... Warm Regards --Himanshu

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[alsa-devel] RESUBSCRIBE - New alsa-devel mailing list
by Jaroslav Kysela 21 Mar '07

21 Mar '07

Hello everyone, because never ending story with SF services, I quickly setup new mailman alsa-devel list on new ALSA server (we'll get working list archive as bonus). Because I don't want to move automatically all subscribers, please, subscribe yourself to new list: http://mailman.alsa-project.org/mailman/listinfo/alsa-devel New list is available as alsa-devel(a)alsa-project.org while old on SF is available as alsa-devel(a)lists.sourceforge.net . The archive for new alsa-devel mailing list is here: http://mailman.alsa-project.org/pipermail/alsa-devel/ The posts from non-members are moderated now! Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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Re: [alsa-devel] Development Question
by M. Gagnon 21 Mar '07

21 Mar '07

> > >What is the output of the program? > > > > pending events size: 28 > > pending events size: 56 > > pending events size: 84 > > ... > > BEFORE DRAIN: pending events size: 476 > > AFTER DRAIN: pending events size: 0 > >Looks OK. In theory, all events should be sent somewhere. > >Please run aseqdump and send the events to its port. What does it >output? > % aseqdump Waiting for data at port 128:0. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 0:1 Port subscribed 129:0 -> 128:0 129:0 Note on 2 60 127 0:1 Port unsubscribed 129:0 -> 128:0 Seems like only one event gets sent - just like what i hear. >Run "cat /proc/asound/seq/queues" and >"cat /proc/asound/seq/timer". > % cat /proc/asound/seq/queues queue 0: [Queue-0] owned by client : 128 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 96 current tempo : 500000 current BPM : 120 current time : 1.536000000 s current tick : 294 queue 1: [Queue-1] owned by client : 129 lock status : Locked queued time events : 0 queued tick events : 0 timer state : Running timer PPQ : 48 current tempo : 1000000 current BPM : 60 current time : 1.488000000 s current tick : 71 % cat /proc/asound/seq/timer Timer for queue 0 : system timer Period time : 0.004000000 Skew : 65536 / 65536 Timer for queue 1 : system timer Period time : 0.004000000 Skew : 65536 / 65536 thanks for your help just as reminder (so the output can be interpreted) this is my latest code /* all: g++ midi_receiver.cpp -o AlsaNoteTest -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> #include "midi_receiver.h" #include <vector> #include <iostream> #include <unistd.h> #include <pthread.h> pthread_t thread1; char *message1 = "Thread 1"; int iret1; void *play_func( void *ptr ) { AlsaNotePlayer::play(); } int main() { AlsaNotePlayer::findDevices(); AlsaNotePlayer::openDevice(128, 0); AlsaNotePlayer::setTempo(60, 48); int time=0; int duration = 50; AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 62, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 62, 0, time); AlsaNotePlayer::noteOn_seq(1, 60, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 60, 0, time); AlsaNotePlayer::noteOn_seq(1, 59, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 59, 0, time); // ---------- AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 65, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 65, 0, time); AlsaNotePlayer::noteOn_seq(1, 63, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 63, 0, time); AlsaNotePlayer::noteOn_seq(1, 61, 127, time); time += duration; AlsaNotePlayer::noteOff_seq(1, 61, 0, time); iret1 = pthread_create( &thread1, NULL, play_func, (void*) message1); pthread_join( thread1, NULL); while(1) { usleep(10000); } } namespace AlsaNotePlayer { class MidiDevice { public: int client, port; char* name; MidiDevice(int client_arg, int port_arg, const char* name_arg) { client = client_arg; port = port_arg; name = (char*)name_arg; } }; std::vector<MidiDevice> devices; struct MidiContext { snd_seq_t* sequencer; int queue; snd_seq_addr_t address; int port; snd_seq_port_subscribe_t *subs; }; struct MidiContext MidiContext; snd_seq_addr_t address; void findDevices() { snd_seq_client_info_t* clientInfo; if (snd_seq_open(&MidiContext.sequencer, "default", SND_SEQ_OPEN_OUTPUT, 0) < 0) { return; } snd_seq_client_info_alloca(&clientInfo); snd_seq_client_info_set_client(clientInfo, -1); // iterate through clients while (snd_seq_query_next_client(MidiContext.sequencer, clientInfo) >= 0) { snd_seq_port_info_t* pinfo; snd_seq_port_info_alloca(&pinfo); snd_seq_port_info_set_client(pinfo, snd_seq_client_info_get_client(clientInfo)); snd_seq_port_info_set_port(pinfo, -1); // and now through ports while (snd_seq_query_next_port(MidiContext.sequencer, pinfo) >= 0) { unsigned int capability = snd_seq_port_info_get_capability(pinfo); if ((capability & SND_SEQ_PORT_CAP_SUBS_WRITE) == 0) { continue; } int client = (snd_seq_port_info_get_addr(pinfo))->client; int port = (snd_seq_port_info_get_addr(pinfo))->port; std::cout << client << ":" << port << " --> " << snd_seq_port_info_get_name(pinfo) << std::endl; devices.push_back( MidiDevice(client, port, snd_seq_port_info_get_name(pinfo)) ); } } if (snd_seq_set_client_name(MidiContext.sequencer, "Aria") < 0) { return; } //MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_APPLICATION); MidiContext.address.port = snd_seq_create_simple_port(MidiContext.sequencer, "Aria Port 0", SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_READ, SND_SEQ_PORT_TYPE_APPLICATION); //SND_SEQ_PORT_TYPE_MIDI_GENERIC); MidiContext.address.client = snd_seq_client_id (MidiContext.sequencer); MidiContext.queue = snd_seq_alloc_queue (MidiContext.sequencer); snd_seq_set_client_pool_output (MidiContext.sequencer, 1024); } void setTempo(int tempo_arg, int resolution) { // argument is not yet considered snd_seq_queue_tempo_t *tempo; snd_seq_queue_tempo_alloca(&tempo); // snd_seq_queue_tempo_set_tempo(tempo, 1000000); // 60 BPM // snd_seq_queue_tempo_set_ppq(tempo, 48); // 48 PPQ snd_seq_queue_tempo_set_ppq(tempo, resolution); snd_seq_queue_tempo_set_tempo(tempo, 60*1000000/tempo_arg); snd_seq_set_queue_tempo(MidiContext.sequencer, MidiContext.queue, tempo); } void openDevice(int client, int port) { address.client = client; address.port = port; snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); snd_seq_subscribe_port(MidiContext.sequencer, MidiContext.subs); } void closeDevice() { snd_seq_port_subscribe_alloca(&MidiContext.subs); snd_seq_port_subscribe_set_sender(MidiContext.subs, &MidiContext.address); snd_seq_port_subscribe_set_dest(MidiContext.subs, &address); if (snd_seq_unsubscribe_port(MidiContext.sequencer, MidiContext.subs) < 0) { return; } //snd_seq_port_subscribe_free(MidiContext.subs); } //************************************************************************************// //************************************MIDI EVENTS*************************************// //************************************************************************************// void noteOn_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_source(&event, MidiContext.port); snd_seq_ev_set_noteon(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void noteOff_seq(int channel, int note, int velocity, int tick) { snd_seq_event_t event; snd_seq_ev_clear(&event); event.queue = MidiContext.queue; event.source = MidiContext.address; snd_seq_ev_set_subs(&event); snd_seq_ev_set_noteoff(&event, channel, note, velocity); snd_seq_ev_schedule_tick(&event, MidiContext.queue, 0 /*absolute*/, tick ); snd_seq_event_output(MidiContext.sequencer, &event); std::cout << "pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; } void play() { snd_seq_start_queue(MidiContext.sequencer, MidiContext.queue, 0); std::cout << "BEFORE DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; int output = snd_seq_drain_output(MidiContext.sequencer); std::cout << "AFTER DRAIN: pending events size: " << snd_seq_event_output_pending(MidiContext.sequencer) << std::endl; if(output == 0) std::cout << "all events drained" << std::endl; if(output > 0) std::cout << output << "bytes remaining" << std::endl; if(output < 0) { fprintf (stderr, "Error: (%s)\n", snd_strerror (output)); //std::cout << "error" << std::endl; } } } _________________________________________________________________ Ne perdez pas de temps dans les files dattente magasinez en ligne. http://magasiner.sympatico.msn.ca

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Re: [alsa-devel] [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards
by Rene Herman 20 Mar '07

20 Mar '07

On 03/19/2007 11:46 PM, Rask Ingemann Lambertsen wrote: >> This setup also shares that same problem with the BIOS generally >> not having reserved/routed the resources -- on a general PCI/ISA >> system the user needs to go into the BIOS setup and reserve the >> resources the card is (or will be, in this case) set to. So now the >> user really needs to know what resources are going to be assigned, >> destroying the plug and play idea... > > Not really. A standards compliant PnPBIOS set up with "PnP aware OS = > YES" will use only enough resources to boot the system, which should > leave enough irqs and dmas for the Linux PnP layer to do its work. There's also the "routed" bit though. I actually ran into this recently when I was playing with snd-sgalaxy (which is why I'm harping on it). An sgalaxy is also a (partly) software configurable card and when not given explicit parameters the driver just searches for an unrequested IRQ and DMA and then programs the card for it. Over here it grabbed IRQ 10 but then didn't work anyway, and only started working when I set IRQ 10 to "Legacy ISA" in the BIOS setup. It seems the IRQ simply was not available to ISA without having done that (no, no PCI cards were using IRQ 10 or anything). I must say though that my comment above was partly based on a somewhat foggy understanding of how things get assigned. Yes, _if_ this "routing" problem I seemed to be experiencing is not a serious problem in practice (Adam? Are you able to comment?) then the fact that the PnP layer will be handing out the resources taking into account the other (known) cards is indeed a practical advantage of using that layer. > I think there are three good reasons to use the PnP layer: > > 1) The PnP layer informs the PCI layer of irqs to try to avoid - please > see pnp_register_irq_resource() in drivers/pnp/resource.c. How does that help? On my systems (one AMI, all others Award BIOses) all PCI cards get assigned their IRQ before the OS ever boots. That "PnP OS: Yes/No" switch only effects whether or not the BIOS will fully configure all ISA-PnP cards. > 2) The PnP layer knows which resources are used by active devices, even > if they have not been allocated using e.g. request_region(). This > happens in practice. For example: Yes, this is an advantage. It does miss all the resources used by old ISA cards for which no (normal or pnp protocol as in this one) drivers have been loaded but I guess that's not much of a difference with a regular ISA driver... > 3) Knowing which resources a card can utilize, the PnP layer can try to > stay away from those resources when configuring other cards. The PnP layer > doesn't do that yet in a stock kernel, but I've patched it to do so. I'll > try to get those patches into the kernel. I guess that could be an advantage. When I was looking at pnp/isa integration a while ago myself I was looking at it from the angle of generic old ISA (ie, hardwired non settable resources) and concluded that for these there wasn't any point. Having looked a little closer at the resource assigment being done in the PnP layer now, I now feel there's some point, but still little for those. But with software settable devices such as this Jazz16, there ofcourse is rather more point. I believe the routing issue I refferred to above is the most important thing to have answered. I also still don't really like "splitting the one driver into the two pieces" like that but well, we don't actually have all that many software settable ISA devices and I guess it's not _too_ bad to have a few of these "front-ends" run as a protocol driver at boot as long as it stays "a few". Hey, I already said I like the concept... By the way; in the "straightforward version" I posted yesterday I neglected to reserve the SB_JAZZ16_CONFIG_PORT and the SB range before poking at it during setup. Attached version corrects that. Updated patch including your SB changes still at http://members.home.nl/rene.herman/jazz16.diff Looks more involved now, but posted mainly for completeness; correcting yesterday's post. Has been tested and found working again. Rene. diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 9fef210..b60690a 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1120,6 +1120,13 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based Media Vision Jazz16 chip + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/include/sound/sb.h b/include/sound/sb.h index 2dd5c8e..9594f89 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -38,6 +38,7 @@ enum sb_hw_type { SB_HW_ALS100, /* Avance Logic ALS100 chip */ SB_HW_ALS4000, /* Avance Logic ALS4000 chip */ SB_HW_DT019X, /* Diamond Tech. DT-019X / Avance Logic ALS-007 */ + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ }; #define SB_OPEN_PCM 0x01 @@ -140,8 +141,11 @@ #define SB_DSP_LO_OUTPUT_AUTO 0x1c #define SB_DSP_LO_INPUT_AUTO 0x2c #define SB_DSP_HI_OUTPUT_AUTO 0x90 #define SB_DSP_HI_INPUT_AUTO 0x98 +#define SB_DSP_ULAW_OUTPUT_AUTO 0x7f +#define SB_DSP_ULAW_INPUT_AUTO 0x8f #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +269,14 @@ #define SB_DMASETUP_DMA5 0x20 #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 + +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index a629b52..450255f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -258,6 +258,14 @@ config SND_SW60XG To compile this driver as a module, choose M here: the module will be called snd-sw60xg. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say 'Y' or 'M' to include support for Media Vision Jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index fd9d9c5..745e1c1 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..988a5e9 --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,334 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * 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 + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * The chip has an MPU-401 interface which is handled by the MPU-401 driver. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <sound/driver.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/ioport.h> +#include <asm/dma.h> +#include <asm/io.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +#define CRD_NAME "Media Vision Jazz16" +#define DEV_NAME "jazz16" + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_LICENSE("GPL"); + +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 long port = SNDRV_DEFAULT_PORT1; /* 0x220,0x240,0x260 */ +static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x310,0x320,0x330 */ +static int irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int mpu_irq = SNDRV_DEFAULT_IRQ1; /* 5,9,3,7,10,15 */ +static int dma8 = SNDRV_DEFAULT_DMA1; /* 1,3 */ +static int dma16 = SNDRV_DEFAULT_DMA1; /* 5,7 */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); +module_param(port, long, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param(mpu_port, long, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param(irq, int, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param(mpu_irq, int, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param(dma8, int, 0444); +MODULE_PARM_DESC(dma8, "8-bit DMA # for " CRD_NAME " driver."); +module_param(dma16, int, 0444); +MODULE_PARM_DESC(dma16, "16-bit DMA # for " CRD_NAME " driver."); + +static int __devinit snd_jazz16_match(struct device *dev, unsigned int n) +{ + if (!enable) + return 0; + + if (port == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + return 0; + } + if (irq == SNDRV_AUTO_IRQ) { + snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + return 0; + } + if (dma8 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma8\n", dev->bus_id); + return 0; + } + if (dma16 == SNDRV_AUTO_DMA) { + snd_printk(KERN_ERR "%s: please specify dma16\n", dev->bus_id); + return 0; + } + return 1; +} + +#define BUSY_LOOPS 100000 + +static int __devinit jazz16_sbdsp_command(u8 value) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { + outb(value, SBP1(port, COMMAND)); + return 0; + } + return -EIO; +} + +static int __devinit jazz16_sbdsp_get_byte(void) +{ + int i; + + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(port, DATA_AVAIL)) & 0x80) + return inb(SBP1(port, READ)); + + return -EIO; +} + +static int __devinit jazz16_sbdsp_reset(void) +{ + int error; + + outb(1, SBP1(port, RESET)); + udelay(10); + outb(0, SBP1(port, RESET)); + udelay(30); + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + return error; + + return error == 0xaa ? 0 : -ENODEV; +} + +static void jazz16_set_ports(u8 value) +{ + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(value, SB_JAZZ16_CONFIG_PORT); +} + +#define JAZZ16_SET_DMAINTR 0xfb + +static int __devinit jazz16_enable(struct device *dev) +{ + static u8 irq_config[] = {0,0,2,3,0,1,0,4,0,2,5,0,0,0,0,6}; + static u8 dma_config[] = {0,1,0,2,0,3,0,4}; + u8 config = 0; + + int error; + + if (port != 0x220 && port != 0x240 && port != 0x260) + port = 0; + + if (mpu_port != 0x310 && mpu_port != 0x320 && mpu_port != 0x330) + mpu_port = 0; + + config = (port & 0x70) | ((mpu_port >> 4) & 0x07); + + snd_printd("%s: setting SB port = %#lx, MPU port = %#lx.\n", + dev->bus_id, port, mpu_port); + + jazz16_set_ports(config); + + snd_printd("%s: setting SB irq %d, dma %d&%d, MPU irq %d.\n", + dev->bus_id, irq, dma8, dma16, mpu_irq); + + if (!request_region(port, 16, "Jazz16")) + return -EBUSY; + + error = jazz16_sbdsp_command(JAZZ16_SET_DMAINTR); + if (error < 0) + goto out; + + config = dma_config[dma8] | dma_config[dma16] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + config = irq_config[irq] | irq_config[mpu_irq] << 4; + + error = jazz16_sbdsp_command(config); + if (error < 0) + goto out; + + error = jazz16_sbdsp_reset(); + if (error < 0) + goto out; + + error = jazz16_sbdsp_command(SB_DSP_GET_JAZZ_VERSION); + if (error < 0) + goto out; + + error = jazz16_sbdsp_get_byte(); + if (error < 0) + goto out; + + error = SB_VERSION_IS_JAZZ16(error) ? 0 : -ENODEV; + +out: release_region(port, 16); + return error; +} + +static irqreturn_t snd_jazz16_interrupt(int irq, void *dev_id) +{ + return snd_sb8dsp_interrupt(dev_id); +} + +static void snd_jazz16_free(struct snd_card *card) +{ + jazz16_set_ports(0); + release_and_free_resource(card->private_data); +} + +static int __devinit snd_jazz16_probe(struct device *dev, unsigned int n) +{ + struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int error; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (!card) + return -EINVAL; + + card->private_data = request_region(SB_JAZZ16_CONFIG_PORT, 1, + "Jazz16 Config"); + if (!card->private_data) { + error = -EBUSY; + goto out; + } + card->private_free = snd_jazz16_free; + + if (mpu_port == SNDRV_AUTO_PORT) + mpu_port = 0; + + if (mpu_irq == SNDRV_AUTO_IRQ) + mpu_irq = 0; + + error = jazz16_enable(dev); + if (error < 0) + goto out; + + error = snd_sbdsp_create(card, port, irq, snd_jazz16_interrupt, + dma8, dma16, SB_HW_AUTO, &chip); + if (error < 0) + goto out; + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(KERN_ERR "%s: not a Jazz16 chip at %#lx.\n", + dev->bus_id, chip->port); + error = -ENODEV; + goto out; + } + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#lx, irq %d, dma8 %d, dma16 %d", + chip->name, chip->port, chip->irq, chip->dma8, chip->dma16); + + error = snd_sb8dsp_pcm(chip, 0, NULL); + if (error < 0) + goto out; + + error = snd_sbmixer_new(chip); + if (error < 0) + goto out; + + error = snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_AUTO, + 1, &opl3); + if (error < 0) + snd_printk(KERN_WARNING "%s: no OPL device found, skipping.\n", + dev->bus_id); + else { + error = snd_opl3_timer_new(opl3, 1, 2); + if (error < 0) + goto out; + + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } + + snd_card_set_dev(card, dev); + + error = snd_card_register(card); + if (error < 0) + goto out; + + dev_set_drvdata(dev, card); + return 0; + +out: snd_card_free(card); + return error; +} + +static int __devexit snd_jazz16_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, 1); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index aea9e5e..70cd031 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sun Mar 11 18:45:38 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements. */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,54 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + unsigned int num = 0, den = 0; + int err = 0; + + if (c->max > 1) + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + else + err = snd_interval_ratnum(r, 1, &clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +169,12 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->playback_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } + /* fall through */ case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +192,26 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +249,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +308,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +320,12 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_MU_LAW) { + chip->capture_format = SB_DSP_ULAW_OUTPUT_AUTO; + break; + } chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +345,26 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + switch (runtime->format) + { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_MU_LAW: + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + break; + + default: + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -278,14 +378,14 @@ static int snd_sb8_capture_prepare(struc } else { snd_sbdsp_command(chip, 256 - runtime->rate_den); } - if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->capture_format != SB_DSP_INPUT) { count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +458,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +471,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +547,20 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_MU_LAW; + runtime->hw.rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -463,9 +581,11 @@ static int snd_sb8_open(struct snd_pcm_s default: break; } + if (chip->hardware == SB_HW_JAZZ16) + return 0; snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); - return 0; + return 0; } static int snd_sb8_close(struct snd_pcm_substream *substream) diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3094f38..69e72b5 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 490b1ca..b16b592 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16:

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[alsa-devel] [Fwd: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards]
by Rene Herman 20 Mar '07

20 Mar '07

I didn't see this one make the old list at all, so forward to the new one... -------- Original Message -------- Subject: Re: [Alsa-devel] [RFC][RFT] Adding support for Jazz16 based sound cards Date: Mon, 19 Mar 2007 23:46:07 +0100 From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> To: Rene Herman <rene.herman(a)gmail.com> CC: alsa-devel(a)lists.sourceforge.net, Adam Belay <ambx1(a)neo.rr.com> References: <20070311232436.GB9778(a)sygehus.dk> <45FDDA0F.7050503(a)gmail.com> On Mon, Mar 19, 2007 at 01:32:15AM +0100, Rene Herman wrote: > > I just now tested this on a standalone Jazz16 and it seems to be working > fine for me. I once had ogg123 segfault on me at the end of playback but > was unsuccesful in reproducing -- may have been a fluke. Thanks. I'll try out ogg123. > As to the comments... > > The changes to the SB code to support the Jazz16 I ofcourse agree with. > Supporting a new card is great. I am however not a huge fan of the new > PnP protocol. It's cute, but I feel it's a little over-engineered and > could just as well live directly in snd-jazz16. See below. > This setup also shares that same problem with the BIOS generally not > having reserved/routed the resources -- on a general PCI/ISA system the > user needs to go into the BIOS setup and reserve the resources the card > is (or will be, in this case) set to. So now the user really needs to > know what resources are going to be assigned, destroying the plug and > play idea... Not really. A standards compliant PnPBIOS set up with "PnP aware OS = YES" will use only enough resources to boot the system, which should leave enough irqs and dmas for the Linux PnP layer to do its work. I think there are three good reasons to use the PnP layer: 1) The PnP layer informs the PCI layer of irqs to try to avoid - please see pnp_register_irq_resource() in drivers/pnp/resource.c. 2) The PnP layer knows which resources are used by active devices, even if they have not been allocated using e.g. request_region(). This happens in practice. For example: $ cat /sys/bus/pnp/devices/01\:02.00/resources state = active io 0x2a0-0x2af io 0x390-0x393 io 0x500-0x50f irq 5 dma 0 dma 1 $ cat /proc/ioports [...] 0200-0200 : ns558-pnp 0201-0201 : ns558-pnp 0208-020f : ns558-pnp 0213-0213 : ISAPnP 0280-028f : ES18xx 0356-0356 : ide2 0360-0361 : MPU401 UART 0376-0376 : 0000:00:07.1 0376-0376 : ide1 0390-0391 : OPL2/3 (left) 0392-0393 : OPL2/3 (right) 03c0-03df : vga+ 03e8-03ef : serial 03f6-03f6 : 0000:00:07.1 03f6-03f6 : ide0 03f8-03ff : serial 0500-050f : AD1816A 0800-0807 : ES18xx - CTRL [...] Notice that 0x2a0-0x2af is missing from /proc/ioports. The PnP layer will avoid using that port range because it knows a device is using it. 3) Knowing which resources a card can utilize, the PnP layer can try to stay away from those resources when configuring other cards. The PnP layer doesn't do that yet in a stock kernel, but I've patched it to do so. I'll try to get those patches into the kernel. An advantage of autoprobing in a PnP protocol driver (as opposed to doing so in a sound card driver) is that it happens before any file systems have been mounted, so if the autoprobe happens to find one of those #%!$@ NE2000 cards and lock up the system, it won't corrupt your disks. -- Rask Ingemann Lambertsen

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[alsa-devel] test1
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test1 ----- Jaroslav Kysela <perex(a)perex.cz>

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[alsa-devel] test
by Jaroslav Kysela 20 Mar '07

20 Mar '07

test ----- Jaroslav Kysela <perex(a)perex.cz>

1 0