audio.c

基于S3CEB2410平台LINUX操作系统下 USB驱动源代码

/*****************************************************************************/

/*
 *	audio.c  --  USB Audio Class driver
 *
 *	Copyright (C) 1999, 2000, 2001
 *	    Alan Cox ([e-mail])
 *	    Thomas Sailer ([e-mail])
 *
 *	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.
 *
 * Debugging:
 * 	Use the 'lsusb' utility to dump the descriptors.
 *
 * 1999-09-07:  Alan Cox
 *		Parsing Audio descriptor patch
 * 1999-09-08:  Thomas Sailer
 *		Added OSS compatible data io functions; both parts of the
 *		driver remain to be glued together
 * 1999-09-10:  Thomas Sailer
 *		Beautified the driver. Added sample format conversions.
 *		Still not properly glued with the parsing code.
 *		The parsing code seems to have its problems btw,
 *		Since it parses all available configs but doesn't
 *		store which iface/altsetting belongs to which config.
 * 1999-09-20:  Thomas Sailer
 *		Threw out Alan's parsing code and implemented my own one.
 *		You cannot reasonnably linearly parse audio descriptors,
 *		especially the AudioClass descriptors have to be considered
 *		pointer lists. Mixer parsing untested, due to lack of device.
 *		First stab at synch pipe implementation, the Dallas USB DAC
 *		wants to use an Asynch out pipe. usb_audio_state now basically
 *		only contains lists of mixer and wave devices. We can therefore
 *		now have multiple mixer/wave devices per USB device.
 * 1999-10-28:  Thomas Sailer
 *		Converted to URB API. Fixed a taskstate/wakeup semantics mistake
 *		that made the driver consume all available CPU cycles.
 *		Now runs stable on UHCI-Acher/Fliegl/Sailer.
 * 1999-10-31:  Thomas Sailer
 *		Audio can now be unloaded if it is not in use by any mixer
 *		or dsp client (formerly you had to disconnect the audio devices
 *		from the USB port)
 *		Finally, about three months after ordering, my "Maxxtro SPK222"
 *		speakers arrived, isn't disdata a great mail order company 8-)
 *		Parse class specific endpoint descriptor of the audiostreaming
 *		interfaces and take the endpoint attributes from there.
 *		Unbelievably, the Philips USB DAC has a sampling rate range
 *		of over a decade, yet does not support the sampling rate control!
 *		No wonder it sounds so bad, has very audible sampling rate
 *		conversion distortion. Don't try to listen to it using
 *		decent headphones!
 *		"Let's make things better" -> but please Philips start with your
 *		own stuff!!!!
 * 1999-11-02:  Thomas Sailer
 *		It takes the Philips boxes several seconds to acquire synchronisation
 *		that means they won't play short sounds. Should probably maintain
 *		the ISO datastream even if there's nothing to play.
 *		Fix counting the total_bytes counter, RealPlayer G2 depends on it.
 * 1999-12-20:  Thomas Sailer
 *		Fix bad bug in conversion to per interface probing.
 *		disconnect was called multiple times for the audio device,
 *		leading to a premature freeing of the audio structures
 * 2000-05-13:  Thomas Sailer
 *		I don't remember who changed the find_format routine,
 *              but the change was completely broken for the Dallas
 *              chip. Anyway taking sampling rate into account in find_format
 *              is bad and should not be done unless there are devices with
 *              completely broken audio descriptors. Unless someone shows
 *              me such a descriptor, I will not allow find_format to
 *              take the sampling rate into account.
 *              Also, the former find_format made:
 *              - mpg123 play mono instead of stereo
 *              - sox completely fail for wav's with sample rates < 44.1kHz
 *                  for the Dallas chip.
 *              Also fix a rather long standing problem with applications that
 *              use "small" writes producing no sound at all.
 * 2000-05-15:  Thomas Sailer
 *		My fears came true, the Philips camera indeed has pretty stupid
 *              audio descriptors.
 * 2000-05-17:  Thomas Sailer
 *		Nemsoft spotted my stupid last minute change, thanks
 * 2000-05-19:  Thomas Sailer
 *		Fixed FEATURE_UNIT thinkos found thanks to the KC Technology
 *              Xtend device. Basically the driver treated FEATURE_UNIT's sourced
 *              by mono terminals as stereo.
 * 2000-05-20:  Thomas Sailer
 *		SELECTOR support (and thus selecting record channels from the mixer).
 *              Somewhat peculiar due to OSS interface limitations. Only works
 *              for channels where a "slider" is already in front of it (i.e.
 *              a MIXER unit or a FEATURE unit with volume capability).
 * 2000-11-26:  Thomas Sailer
 *              Workaround for Dallas DS4201. The DS4201 uses PCM8 as format tag for
 *              its 8 bit modes, but expects signed data (and should therefore have used PCM).
 * 2001-03-10:  Thomas Sailer
 *              provide abs function, prevent picking up a bogus kernel macro
 *              for abs. Bug report by Andrew Morton <[e-mail]>
 * 2001-06-16:  Bryce Nesbitt <[e-mail]>
 *              Fix SNDCTL_DSP_STEREO API violation
 */

/*
 * Strategy:
 *
 * Alan Cox and Thomas Sailer are starting to dig at opposite ends and
 * are hoping to meet in the middle, just like tunnel diggers :)
 * Alan tackles the descriptor parsing, Thomas the actual data IO and the
 * OSS compatible interface.
 *
 * Data IO implementation issues
 *
 * A mmap'able ring buffer per direction is implemented, because
 * almost every OSS app expects it. It is however impractical to
 * transmit/receive USB data directly into and out of the ring buffer,
 * due to alignment and synchronisation issues. Instead, the ring buffer
 * feeds a constant time delay line that handles the USB issues.
 *
 * Now we first try to find an alternate setting that exactly matches
 * the sample format requested by the user. If we find one, we do not
 * need to perform any sample rate conversions. If there is no matching
 * altsetting, we choose the closest one and perform sample format
 * conversions. We never do sample rate conversion; these are too
 * expensive to be performed in the kernel.
 *
 * Current status:
 * - Pretty stable on UHCI-Acher/Fliegl/Sailer
 * - Does not work on OHCI due to lack of OHCI driver supporting URB's
 *
 * Generally: Due to the brokenness of the Audio Class spec
 * it seems generally impossible to write a generic Audio Class driver,
 * so a reasonable driver should implement the features that are actually
 * used.
 *
 * Parsing implementation issues
 *
 * One cannot reasonably parse the AudioClass descriptors linearly.
 * Therefore the current implementation features routines to look
 * for a specific descriptor in the descriptor list.
 *
 * How does the parsing work? First, all interfaces are searched
 * for an AudioControl class interface. If found, the config descriptor
 * that belongs to the current configuration is fetched from the device.
 * Then the HEADER descriptor is fetched. It contains a list of
 * all AudioStreaming and MIDIStreaming devices. This list is then walked,
 * and all AudioStreaming interfaces are classified into input and output
 * interfaces (according to the endpoint0 direction in altsetting1) (MIDIStreaming
 * is currently not supported). The input & output list is then used
 * to group inputs and outputs together and issued pairwise to the
 * AudioStreaming class parser. Finally, all OUTPUT_TERMINAL descriptors
 * are walked and issued to the mixer construction routine.
 *
 * The AudioStreaming parser simply enumerates all altsettings belonging
 * to the specified interface. It looks for AS_GENERAL and FORMAT_TYPE
 * class specific descriptors to extract the sample format/sample rate
 * data. Only sample format types PCM and PCM8 are supported right now, and
 * only FORMAT_TYPE_I is handled. The isochronous data endpoint needs to
 * be the first endpoint of the interface, and the optional synchronisation
 * isochronous endpoint the second one.
 *
 * Mixer construction works as follows: The various TERMINAL and UNIT
 * descriptors span a tree from the root (OUTPUT_TERMINAL) through the
 * intermediate nodes (UNITs) to the leaves (INPUT_TERMINAL). We walk
 * that tree in a depth first manner. FEATURE_UNITs may contribute volume,
 * bass and treble sliders to the mixer, MIXER_UNITs volume sliders.
 * The terminal type encoded in the INPUT_TERMINALs feeds a heuristic
 * to determine "meaningful" OSS slider numbers, however we will see
 * how well this works in practice. Other features are not used at the
 * moment, they seem less often used. Also, it seems difficult at least
 * to construct recording source switches from SELECTOR_UNITs, but
 * since there are not many USB ADC's available, we leave that for later.
 */

/*****************************************************************************/

#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/slab.h>
#include <linux/string.h>
#include <linux/timer.h>
#include <linux/sched.h>
#include <linux/smp_lock.h>
#include <linux/module.h>
#include <linux/sound.h>
#include <linux/soundcard.h>
#include <linux/list.h>
#include <linux/vmalloc.h>
#include <linux/wrapper.h>
#include <linux/init.h>
#include <linux/poll.h>
#include <linux/bitops.h>
#include <asm/uaccess.h>
#include <asm/io.h>
#include <linux/usb.h>

#include "audio.h"

/*
 * Version Information
 */
#define DRIVER_VERSION "v1.0.0"
#define DRIVER_AUTHOR "Alan Cox <[e-mail]>, Thomas Sailer ([e-mail])"
#define DRIVER_DESC "USB Audio Class driver"

#define AUDIO_DEBUG 1

#define SND_DEV_DSP16   5 

#define dprintk(x)

#undef abs
extern int abs(int __x) __attribute__ ((__const__)); /* Shut up warning */

/* --------------------------------------------------------------------- */

/*
 * Linked list of all audio devices...
 */
static struct list_head audiodevs = LIST_HEAD_INIT(audiodevs);
static DECLARE_MUTEX(open_sem);

/*
 * wait queue for processes wanting to open an USB audio device
 */
static DECLARE_WAIT_QUEUE_HEAD(open_wait);


#define MAXFORMATS        MAX_ALT
#define DMABUFSHIFT       17  /* 128k worth of DMA buffer */
#define NRSGBUF           (1U<<(DMABUFSHIFT-PAGE_SHIFT))

/*
 * This influences:
 * - Latency
 * - Interrupt rate
 * - Synchronisation behaviour
 * Don't touch this if you don't understand all of the above.
 */
#define DESCFRAMES  5
#define SYNCFRAMES  DESCFRAMES

#define MIXFLG_STEREOIN   1
#define MIXFLG_STEREOOUT  2

struct mixerchannel {
	__u16 value;
	__u16 osschannel;  /* number of the OSS channel */
	__s16 minval, maxval;
	__u16 slctunitid;
	__u8 unitid;
	__u8 selector;
	__u8 chnum;
	__u8 flags;
};

struct audioformat {
	unsigned int format;
	unsigned int sratelo;
	unsigned int sratehi;
	unsigned char altsetting;
	unsigned char attributes;
};

struct dmabuf {
	/* buffer data format */
	unsigned int format;
	unsigned int srate;
	/* physical buffer */
	unsigned char *sgbuf[NRSGBUF];
	unsigned bufsize;
	unsigned numfrag;
	unsigned fragshift;
	unsigned wrptr, rdptr;
	unsigned total_bytes;
	int count;
	unsigned error; /* over/underrun */
	wait_queue_head_t wait;
	/* redundant, but makes calculations easier */
	unsigned fragsize;
	unsigned dmasize;
	/* OSS stuff */
	unsigned mapped:1;
	unsigned ready:1;
	unsigned ossfragshift;
	int ossmaxfrags;
	unsigned subdivision;
};

struct usb_audio_state;

#define FLG_URB0RUNNING   1
#define FLG_URB1RUNNING   2
#define FLG_SYNC0RUNNING  4
#define FLG_SYNC1RUNNING  8
#define FLG_RUNNING      16
#define FLG_CONNECTED    32

struct my_data_urb {
	urb_t urb;
	iso_packet_descriptor_t isoframe[DESCFRAMES];
};

struct my_sync_urb {
	urb_t urb;
	iso_packet_descriptor_t isoframe[SYNCFRAMES];
};


struct usb_audiodev {
	struct list_head list;
	struct usb_audio_state *state;
	
	/* soundcore stuff */
	int dev_audio;

	/* wave stuff */
	mode_t open_mode;
	spinlock_t lock;         /* DMA buffer access spinlock */

	struct usbin {
		int interface;           /* Interface number, -1 means not used */
		unsigned int format;     /* USB data format */
		unsigned int datapipe;   /* the data input pipe */
		unsigned int syncpipe;   /* the synchronisation pipe - 0 for anything but adaptive IN mode */
		unsigned int syncinterval;  /* P for adaptive IN mode, 0 otherwise */
		unsigned int freqn;      /* nominal sampling rate in USB format, i.e. fs/1000 in Q10.14 */
		unsigned int freqmax;    /* maximum sampling rate, used for buffer management */
		unsigned int phase;      /* phase accumulator */
		unsigned int flags;      /* see FLG_ defines */
		
		struct my_data_urb durb[2];  /* ISO descriptors for the data endpoint */
		struct my_sync_urb surb[2];  /* ISO sync pipe descriptor if needed */
		
		struct dmabuf dma;
	} usbin;

	struct usbout {
		int interface;           /* Interface number, -1 means not used */
		unsigned int format;     /* USB data format */
		unsigned int datapipe;   /* the data input pipe */
		unsigned int syncpipe;   /* the synchronisation pipe - 0 for anything but asynchronous OUT mode */
		unsigned int syncinterval;  /* P for asynchronous OUT mode, 0 otherwise */
		unsigned int freqn;      /* nominal sampling rate in USB format, i.e. fs/1000 in Q10.14 */
		unsigned int freqm;      /* momentary sampling rate in USB format, i.e. fs/1000 in Q10.14 */
		unsigned int freqmax;    /* maximum sampling rate, used for buffer management */
		unsigned int phase;      /* phase accumulator */
		unsigned int flags;      /* see FLG_ defines */

		struct my_data_urb durb[2];  /* ISO descriptors for the data endpoint */
		struct my_sync_urb surb[2];  /* ISO sync pipe descriptor if needed */
		
		struct dmabuf dma;
	} usbout;


	unsigned int numfmtin, numfmtout;
	struct audioformat fmtin[MAXFORMATS];
	struct audioformat fmtout[MAXFORMATS];
};  

struct usb_mixerdev {
	struct list_head list;
	struct usb_audio_state *state;

	/* soundcore stuff */
	int dev_mixer;

	unsigned char iface;  /* interface number of the AudioControl interface */

	/* USB format descriptions */
	unsigned int numch, modcnt;

	/* mixch is last and gets allocated dynamically */
	struct mixerchannel ch[0];
};

struct usb_audio_state {
	struct list_head audiodev;

	/* USB device */
	struct usb_device *usbdev;

	struct list_head audiolist;
	struct list_head mixerlist;

	unsigned count;  /* usage counter; NOTE: the usb stack is also considered a user */
};

/* private audio format extensions */
#define AFMT_STEREO        0x80000000
#define AFMT_ISSTEREO(x)   ((x) & AFMT_STEREO)
#define AFMT_IS16BIT(x)    ((x) & (AFMT_S16_LE|AFMT_S16_BE|AFMT_U16_LE|AFMT_U16_BE))
#define AFMT_ISUNSIGNED(x) ((x) & (AFMT_U8|AFMT_U16_LE|AFMT_U16_BE))
#define AFMT_BYTESSHIFT(x) ((AFMT_ISSTEREO(x) ? 1 : 0) + (AFMT_IS16BIT(x) ? 1 : 0))
#define AFMT_BYTES(x)      (1<<AFMT_BYTESSHFIT(x))

/* --------------------------------------------------------------------- */

/* prevent picking up a bogus abs macro */
#undef abs
extern inline int abs(int x)
{
        if (x < 0)
		return -x;
	return x;
}
                                
/* --------------------------------------------------------------------- */

extern inline unsigned ld2(unsigned int x)
{
	unsigned r = 0;
	
	if (x >= 0x10000) {
		x >>= 16;
		r += 16;
	}