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v5.9
  1/* SPDX-License-Identifier: GPL-2.0-or-later */
  2/*
  3 *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
  4 *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
  5 *  Version: 0.0.22
  6 *
  7 *  FEATURES currently supported:
  8 *    See ca0106_main.c for features.
  9 * 
 10 *  Changelog:
 11 *    Support interrupts per period.
 12 *    Removed noise from Center/LFE channel when in Analog mode.
 13 *    Rename and remove mixer controls.
 14 *  0.0.6
 15 *    Use separate card based DMA buffer for periods table list.
 16 *  0.0.7
 17 *    Change remove and rename ctrls into lists.
 18 *  0.0.8
 19 *    Try to fix capture sources.
 20 *  0.0.9
 21 *    Fix AC3 output.
 22 *    Enable S32_LE format support.
 23 *  0.0.10
 24 *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
 25 *  0.0.11
 26 *    Add Model name recognition.
 27 *  0.0.12
 28 *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
 29 *    Remove redundent "voice" handling.
 30 *  0.0.13
 31 *    Single trigger call for multi channels.
 32 *  0.0.14
 33 *    Set limits based on what the sound card hardware can do.
 34 *    playback periods_min=2, periods_max=8
 35 *    capture hw constraints require period_size = n * 64 bytes.
 36 *    playback hw constraints require period_size = n * 64 bytes.
 37 *  0.0.15
 38 *    Separated ca0106.c into separate functional .c files.
 39 *  0.0.16
 40 *    Implement 192000 sample rate.
 41 *  0.0.17
 42 *    Add support for SB0410 and SB0413.
 43 *  0.0.18
 44 *    Modified Copyright message.
 45 *  0.0.19
 46 *    Added I2C and SPI registers. Filled in interrupt enable.
 47 *  0.0.20
 48 *    Added GPIO info for SB Live 24bit.
 49 *  0.0.21
 50 *   Implement support for Line-in capture on SB Live 24bit.
 51 *  0.0.22
 52 *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
 53 *
 
 54 *  This code was initially based on code from ALSA's emu10k1x.c which is:
 55 *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 56 */
 57
 58/************************************************************************************************/
 59/* PCI function 0 registers, address = <val> + PCIBASE0						*/
 60/************************************************************************************************/
 61
 62#define PTR			0x00		/* Indexed register set pointer register	*/
 63						/* NOTE: The CHANNELNUM and ADDRESS words can	*/
 64						/* be modified independently of each other.	*/
 65						/* CNL[1:0], ADDR[27:16]                        */
 66
 67#define DATA			0x04		/* Indexed register set data register		*/
 68						/* DATA[31:0]					*/
 69
 70#define IPR			0x08		/* Global interrupt pending register		*/
 71						/* Clear pending interrupts by writing a 1 to	*/
 72						/* the relevant bits and zero to the other bits	*/
 73#define IPR_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
 74#define IPR_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
 75#define IPR_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
 76#define IPR_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
 77#define IPR_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
 78#define IPR_SPI			0x00000800      /* SPI transaction completed			*/
 79#define IPR_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
 80#define IPR_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
 81#define IPR_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x76	*/
 82#define IPR_GPI			0x00000080      /* General Purpose input changed		*/
 83#define IPR_SRC_LOCKED          0x00000040      /* SRC lock status changed			*/
 84#define IPR_SPDIF_STATUS        0x00000020      /* SPDIF status changed				*/
 85#define IPR_TIMER2              0x00000010      /* 192000Hz Timer				*/
 86#define IPR_TIMER1              0x00000008      /* 44100Hz Timer				*/
 87#define IPR_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
 88#define IPR_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
 89#define IPR_PCI			0x00000001	/* PCI Bus error				*/
 90
 91#define INTE			0x0c		/* Interrupt enable register			*/
 92
 93#define INTE_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
 94#define INTE_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
 95#define INTE_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
 96#define INTE_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
 97#define INTE_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
 98#define INTE_SPI		0x00000800      /* SPI transaction completed			*/
 99#define INTE_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
100#define INTE_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
101#define INTE_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x75 */
102#define INTE_GPI		0x00000080      /* General Purpose input changed		*/
103#define INTE_SRC_LOCKED         0x00000040      /* SRC lock status changed			*/
104#define INTE_SPDIF_STATUS       0x00000020      /* SPDIF status changed				*/
105#define INTE_TIMER2             0x00000010      /* 192000Hz Timer				*/
106#define INTE_TIMER1             0x00000008      /* 44100Hz Timer				*/
107#define INTE_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
108#define INTE_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
109#define INTE_PCI		0x00000001	/* PCI Bus error				*/
110
111#define UNKNOWN10		0x10		/* Unknown ??. Defaults to 0 */
112#define HCFG			0x14		/* Hardware config register			*/
113						/* 0x1000 causes AC3 to fails. It adds a dither bit. */
114
115#define HCFG_STAC		0x10000000	/* Special mode for STAC9460 Codec. */
116#define HCFG_CAPTURE_I2S_BYPASS	0x08000000	/* 1 = bypass I2S input async SRC. */
117#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000	/* 1 = bypass SPDIF input async SRC. */
118#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000	/* 0 = I2S IN mixer output, 1 = I2S IN1. */
119#define HCFG_FORCE_LOCK		0x01000000	/* For test only. Force input SRC tracker to lock. */
120#define HCFG_PLAYBACK_ATTENUATION 0x00006000	/* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
121#define HCFG_PLAYBACK_DITHER	0x00001000	/* 1 = Add dither bit to all playback channels. */
122#define HCFG_PLAYBACK_S32_LE	0x00000800	/* 1 = S32_LE, 0 = S16_LE                       */
123#define HCFG_CAPTURE_S32_LE	0x00000400	/* 1 = S32_LE, 0 = S16_LE (S32_LE current not working)	*/
124#define HCFG_8_CHANNEL_PLAY	0x00000200	/* 1 = 8 channels, 0 = 2 channels per substream.*/
125#define HCFG_8_CHANNEL_CAPTURE	0x00000100	/* 1 = 8 channels, 0 = 2 channels per substream.*/
126#define HCFG_MONO		0x00000080	/* 1 = I2S Input mono                           */
127#define HCFG_I2S_OUTPUT		0x00000010	/* 1 = I2S Output disabled                      */
128#define HCFG_AC97		0x00000008	/* 0 = AC97 1.0, 1 = AC97 2.0                   */
129#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004	/* 1 = Cancel bustmaster accesses to soundcache */
130						/* NOTE: This should generally never be used.  	*/
131#define HCFG_LOCK_CAPTURE_CACHE	0x00000002	/* 1 = Cancel bustmaster accesses to soundcache */
132						/* NOTE: This should generally never be used.  	*/
133#define HCFG_AUDIOENABLE	0x00000001	/* 0 = CODECs transmit zero-valued samples	*/
134						/* Should be set to 1 when the EMU10K1 is	*/
135						/* completely initialized.			*/
136#define GPIO			0x18		/* Defaults: 005f03a3-Analog, 005f02a2-SPDIF.   */
137						/* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
138						/* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
139						/* SB Live 24bit:
140						 * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
141						 * bit 9 0 = Mute / 1 = Analog out.
142						 * bit 10 0 = Line-in / 1 = Mic-in.
143						 * bit 11 0 = ? / 1 = ?
144						 * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
145						 * bit 13 0 = ? / 1 = ?
146						 * bit 14 0 = Mute / 1 = Analog out
147						 * bit 15 0 = ? / 1 = ?
148						 * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
149						 */
150						/* 8 general purpose programmable In/Out pins.
151						 * GPI [8:0] Read only. Default 0.
152						 * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
153						 * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
154						 */
155#define AC97DATA		0x1c		/* AC97 register set data register (16 bit)	*/
156
157#define AC97ADDRESS		0x1e		/* AC97 register set address register (8 bit)	*/
158
159/********************************************************************************************************/
160/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
161/********************************************************************************************************/
162                                                                                                                           
163/* Initially all registers from 0x00 to 0x3f have zero contents. */
164#define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
165						/* One list entry: 4 bytes for DMA address, 
166						 * 4 bytes for period_size << 16.
167						 * One list entry is 8 bytes long.
168						 * One list entry for each period in the buffer.
169						 */
170						/* ADDR[31:0], Default: 0x0 */
171#define PLAYBACK_LIST_SIZE	0x01		/* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000  */
172						/* SIZE[21:16], Default: 0x8 */
173#define PLAYBACK_LIST_PTR	0x02		/* Pointer to the current period being played */
174						/* PTR[5:0], Default: 0x0 */
175#define PLAYBACK_UNKNOWN3	0x03		/* Not used ?? */
176#define PLAYBACK_DMA_ADDR	0x04		/* Playback DMA address */
177						/* DMA[31:0], Default: 0x0 */
178#define PLAYBACK_PERIOD_SIZE	0x05		/* Playback period size. win2000 uses 0x04000000 */
179						/* SIZE[31:16], Default: 0x0 */
180#define PLAYBACK_POINTER	0x06		/* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
181						/* POINTER[15:0], Default: 0x0 */
182#define PLAYBACK_PERIOD_END_ADDR 0x07		/* Playback fifo end address */
183						/* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
184#define PLAYBACK_FIFO_OFFSET_ADDRESS	0x08	/* Current fifo offset address [21:16] */
185						/* Cache size valid [5:0] */
186#define PLAYBACK_UNKNOWN9	0x09		/* 0x9 to 0xf Unused */
187#define CAPTURE_DMA_ADDR	0x10		/* Capture DMA address */
188						/* DMA[31:0], Default: 0x0 */
189#define CAPTURE_BUFFER_SIZE	0x11		/* Capture buffer size */
190						/* SIZE[31:16], Default: 0x0 */
191#define CAPTURE_POINTER		0x12		/* Capture buffer pointer. Sample currently in ADC */
192						/* POINTER[15:0], Default: 0x0 */
193#define CAPTURE_FIFO_OFFSET_ADDRESS	0x13	/* Current fifo offset address [21:16] */
194						/* Cache size valid [5:0] */
195#define PLAYBACK_LAST_SAMPLE    0x20		/* The sample currently being played */
196/* 0x21 - 0x3f unused */
197#define BASIC_INTERRUPT         0x40		/* Used by both playback and capture interrupt handler */
198						/* Playback (0x1<<channel_id) */
199						/* Capture  (0x100<<channel_id) */
200						/* Playback sample rate 96000 = 0x20000 */
201						/* Start Playback [3:0] (one bit per channel)
202						 * Start Capture [11:8] (one bit per channel)
203						 * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
204						 * Playback mixer in enable [27:24] (one bit per channel)
205						 * Playback mixer out enable [31:28] (one bit per channel)
206						 */
207/* The Digital out jack is shared with the Center/LFE Analogue output. 
208 * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
209 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
210 * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
211 * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
212 * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
213 */
214/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
215 * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
216 * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
217 * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
218 */
219/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
220 * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. 
221 */
222#define SPCS0			0x41		/* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006	*/
223#define SPCS1			0x42		/* SPDIF output Channel Status 1 register. For Front */
224#define SPCS2			0x43		/* SPDIF output Channel Status 2 register. For Center/LFE */
225#define SPCS3			0x44		/* SPDIF output Channel Status 3 register. Unknown */
226						/* When Channel set to 0: */
227#define SPCS_CLKACCYMASK	0x30000000	/* Clock accuracy				*/
228#define SPCS_CLKACCY_1000PPM	0x00000000	/* 1000 parts per million			*/
229#define SPCS_CLKACCY_50PPM	0x10000000	/* 50 parts per million				*/
230#define SPCS_CLKACCY_VARIABLE	0x20000000	/* Variable accuracy				*/
231#define SPCS_SAMPLERATEMASK	0x0f000000	/* Sample rate					*/
232#define SPCS_SAMPLERATE_44	0x00000000	/* 44.1kHz sample rate				*/
233#define SPCS_SAMPLERATE_48	0x02000000	/* 48kHz sample rate				*/
234#define SPCS_SAMPLERATE_32	0x03000000	/* 32kHz sample rate				*/
235#define SPCS_CHANNELNUMMASK	0x00f00000	/* Channel number				*/
236#define SPCS_CHANNELNUM_UNSPEC	0x00000000	/* Unspecified channel number			*/
237#define SPCS_CHANNELNUM_LEFT	0x00100000	/* Left channel					*/
238#define SPCS_CHANNELNUM_RIGHT	0x00200000	/* Right channel				*/
239#define SPCS_SOURCENUMMASK	0x000f0000	/* Source number				*/
240#define SPCS_SOURCENUM_UNSPEC	0x00000000	/* Unspecified source number			*/
241#define SPCS_GENERATIONSTATUS	0x00008000	/* Originality flag (see IEC-958 spec)		*/
242#define SPCS_CATEGORYCODEMASK	0x00007f00	/* Category code (see IEC-958 spec)		*/
243#define SPCS_MODEMASK		0x000000c0	/* Mode (see IEC-958 spec)			*/
244#define SPCS_EMPHASISMASK	0x00000038	/* Emphasis					*/
245#define SPCS_EMPHASIS_NONE	0x00000000	/* No emphasis					*/
246#define SPCS_EMPHASIS_50_15	0x00000008	/* 50/15 usec 2 channel				*/
247#define SPCS_COPYRIGHT		0x00000004	/* Copyright asserted flag -- do not modify	*/
248#define SPCS_NOTAUDIODATA	0x00000002	/* 0 = Digital audio, 1 = not audio		*/
249#define SPCS_PROFESSIONAL	0x00000001	/* 0 = Consumer (IEC-958), 1 = pro (AES3-1992)	*/
250
251						/* When Channel set to 1: */
252#define SPCS_WORD_LENGTH_MASK	0x0000000f	/* Word Length Mask				*/
253#define SPCS_WORD_LENGTH_16	0x00000008	/* Word Length 16 bit				*/
254#define SPCS_WORD_LENGTH_17	0x00000006	/* Word Length 17 bit				*/
255#define SPCS_WORD_LENGTH_18	0x00000004	/* Word Length 18 bit				*/
256#define SPCS_WORD_LENGTH_19	0x00000002	/* Word Length 19 bit				*/
257#define SPCS_WORD_LENGTH_20A	0x0000000a	/* Word Length 20 bit				*/
258#define SPCS_WORD_LENGTH_20	0x00000009	/* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
259#define SPCS_WORD_LENGTH_21	0x00000007	/* Word Length 21 bit				*/
260#define SPCS_WORD_LENGTH_22	0x00000005	/* Word Length 22 bit				*/
261#define SPCS_WORD_LENGTH_23	0x00000003	/* Word Length 23 bit				*/
262#define SPCS_WORD_LENGTH_24	0x0000000b	/* Word Length 24 bit				*/
263#define SPCS_ORIGINAL_SAMPLE_RATE_MASK	0x000000f0 /* Original Sample rate			*/
264#define SPCS_ORIGINAL_SAMPLE_RATE_NONE	0x00000000 /* Original Sample rate not indicated	*/
265#define SPCS_ORIGINAL_SAMPLE_RATE_16000	0x00000010 /* Original Sample rate	*/
266#define SPCS_ORIGINAL_SAMPLE_RATE_RES1	0x00000020 /* Original Sample rate	*/
267#define SPCS_ORIGINAL_SAMPLE_RATE_32000	0x00000030 /* Original Sample rate	*/
268#define SPCS_ORIGINAL_SAMPLE_RATE_12000	0x00000040 /* Original Sample rate	*/
269#define SPCS_ORIGINAL_SAMPLE_RATE_11025	0x00000050 /* Original Sample rate	*/
270#define SPCS_ORIGINAL_SAMPLE_RATE_8000	0x00000060 /* Original Sample rate	*/
271#define SPCS_ORIGINAL_SAMPLE_RATE_RES2	0x00000070 /* Original Sample rate	*/
272#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate	*/
273#define SPCS_ORIGINAL_SAMPLE_RATE_24000	0x00000090 /* Original Sample rate	*/
274#define SPCS_ORIGINAL_SAMPLE_RATE_96000	0x000000a0 /* Original Sample rate	*/
275#define SPCS_ORIGINAL_SAMPLE_RATE_48000	0x000000b0 /* Original Sample rate	*/
276#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate	*/
277#define SPCS_ORIGINAL_SAMPLE_RATE_22050	0x000000d0 /* Original Sample rate	*/
278#define SPCS_ORIGINAL_SAMPLE_RATE_88200	0x000000e0 /* Original Sample rate	*/
279#define SPCS_ORIGINAL_SAMPLE_RATE_44100	0x000000f0 /* Original Sample rate	*/
280
281#define SPDIF_SELECT1		0x45		/* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
282						/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
283						 * But as the jack is shared, use 0xf00.
284						 * The Windows2000 driver uses 0x0000000f for both digital and analog.
285						 * 0xf00 introduces interesting noises onto the Center/LFE.
286						 * If you turn the volume up, you hear computer noise,
287						 * e.g. mouse moving, changing between app windows etc.
288						 * So, I am going to set this to 0x0000000f all the time now,
289						 * same as the windows driver does.
290						 * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
291						 */
292						/* When Channel = 0:
293						 * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
294						 * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
295						 * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
296						 */
297						/* When Channel = 1:
298						 * SPDIF 0 User data [7:0]
299						 * SPDIF 1 User data [15:8]
300						 * SPDIF 0 User data [23:16]
301						 * SPDIF 0 User data [31:24]
302						 * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
303						 */
304#define WATERMARK		0x46		/* Test bit to indicate cache usage level */
305#define SPDIF_INPUT_STATUS	0x49		/* SPDIF Input status register. Bits the same as SPCS.
306						 * When Channel = 0: Bits the same as SPCS channel 0.
307						 * When Channel = 1: Bits the same as SPCS channel 1.
308						 * When Channel = 2:
309						 * SPDIF Input User data [16:0]
310						 * SPDIF Input Frame count [21:16]
311						 */
312#define CAPTURE_CACHE_DATA	0x50		/* 0x50-0x5f Recorded samples. */
313#define CAPTURE_SOURCE          0x60            /* Capture Source 0 = MIC */
314#define CAPTURE_SOURCE_CHANNEL0 0xf0000000	/* Mask for selecting the Capture sources */
315#define CAPTURE_SOURCE_CHANNEL1 0x0f000000	/* 0 - SPDIF mixer output. */
316#define CAPTURE_SOURCE_CHANNEL2 0x00f00000      /* 1 - What you hear or . 2 - ?? */
317#define CAPTURE_SOURCE_CHANNEL3 0x000f0000	/* 3 - Mic in, Line in, TAD in, Aux in. */
318#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff	/* Default 0x00e4 */
319						/* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 
320						 * Record source select for channel 0 [18:16]
321						 * Record source select for channel 1 [22:20]
322						 * Record source select for channel 2 [26:24]
323						 * Record source select for channel 3 [30:28]
324						 * 0 - SPDIF mixer output.
325						 * 1 - i2s mixer output.
326						 * 2 - SPDIF input.
327						 * 3 - i2s input.
328						 * 4 - AC97 capture.
329						 * 5 - SRC output.
330						 */
331#define CAPTURE_VOLUME1         0x61            /* Capture  volume per channel 0-3 */
332#define CAPTURE_VOLUME2         0x62            /* Capture  volume per channel 4-7 */
333
334#define PLAYBACK_ROUTING1       0x63            /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
335#define ROUTING1_REAR           0x77000000      /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
336#define ROUTING1_NULL           0x00770000      /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
337#define ROUTING1_CENTER_LFE     0x00007700      /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
338#define ROUTING1_FRONT          0x00000077	/* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
339						/* Channel_id's handle stereo channels. Channel X is a single mono channel */
340						/* Host is input from the PCI bus. */
341						/* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
342						 * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
343						 * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
344						 * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
345						 * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
346						 * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
347						 * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
348						 * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
349						 */
350
351#define PLAYBACK_ROUTING2       0x64            /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
352						/* SRC is input from the capture inputs. */
353						/* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
354						 * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
355						 * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
356						 * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
357						 * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
358						 * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
359						 * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
360						 * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
361						 */
362
363#define PLAYBACK_MUTE           0x65            /* Unknown. While playing 0x0, while silent 0x00fc0000 */
364						/* SPDIF Mixer input control:
365						 * Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
366						 * Invert Host to SPDIF Mixer [15:8] (One bit per channel)
367						 * SRC to SPDIF Mixer disable [23:16] (One bit per channel)
368						 * Host to SPDIF Mixer disable [31:24] (One bit per channel)
369						 */
370#define PLAYBACK_VOLUME1        0x66            /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
371						/* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
372						/* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
373						/* One register for each of the 4 stereo streams. */
374						/* SRC Right volume [7:0]
375						 * SRC Left  volume [15:8]
376						 * Host Right volume [23:16]
377						 * Host Left  volume [31:24]
378						 */
379#define CAPTURE_ROUTING1        0x67            /* Capture Routing. Default 0x32765410 */
380						/* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
381#define CAPTURE_ROUTING2        0x68            /* Unknown Routing. Default 0x76767676 */
382						/* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
383#define CAPTURE_MUTE            0x69            /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
384						/* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
385#define PLAYBACK_VOLUME2        0x6a            /* Playback Analog volume per channel. Does not effect AC3 output */
386						/* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
387#define UNKNOWN6b               0x6b            /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
388#define MIDI_UART_A_DATA		0x6c            /* Midi Uart A Data */
389#define MIDI_UART_A_CMD		0x6d            /* Midi Uart A Command/Status */
390#define MIDI_UART_B_DATA		0x6e            /* Midi Uart B Data (currently unused) */
391#define MIDI_UART_B_CMD		0x6f            /* Midi Uart B Command/Status (currently unused) */
392
393/* unique channel identifier for midi->channel */
394
395#define CA0106_MIDI_CHAN_A		0x1
396#define CA0106_MIDI_CHAN_B		0x2
397
398/* from mpu401 */
399
400#define CA0106_MIDI_INPUT_AVAIL 	0x80
401#define CA0106_MIDI_OUTPUT_READY	0x40
402#define CA0106_MPU401_RESET		0xff
403#define CA0106_MPU401_ENTER_UART	0x3f
404#define CA0106_MPU401_ACK		0xfe
405
406#define SAMPLE_RATE_TRACKER_STATUS 0x70         /* Readonly. Default 00108000 00108000 00500000 00500000 */
407						/* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 =  1.0
408						 * Rate Locked [20]
409						 * SPDIF Locked [21] For SPDIF channel only.
410						 * Valid Audio [22] For SPDIF channel only.
411						 */
412#define CAPTURE_CONTROL         0x71            /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
413						/* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
414						/* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
415						/* Sample rate output control register Channel=0
416						 * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
417						 * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
418						 * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
419						 * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
420						 * Record mixer output enable [12:10] 
421						 * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
422						 * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
423						 * I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
424						 * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
425						 * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
426						 * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
427						 * I2S input mode [23] (0=Slave, 1=Master)
428						 * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
429						 * SPDIF output source select [26] (0=host, 1=SRC)
430						 * Not used [27]
431						 * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
432						 * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
433						 */ 
434						/* Sample rate output control register Channel=1
435						 * I2S Input 0 volume Right [7:0]
436						 * I2S Input 0 volume Left [15:8]
437						 * I2S Input 1 volume Right [23:16]
438						 * I2S Input 1 volume Left [31:24]
439						 */
440						/* Sample rate output control register Channel=2
441						 * SPDIF Input volume Right [23:16]
442						 * SPDIF Input volume Left [31:24]
443						 */
444						/* Sample rate output control register Channel=3
445						 * No used
446						 */
447#define SPDIF_SELECT2           0x72            /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
448#define ROUTING2_FRONT_MASK     0x00010000      /* Enable for Front speakers. */
449#define ROUTING2_CENTER_LFE_MASK 0x00020000     /* Enable for Center/LFE speakers. */
450#define ROUTING2_REAR_MASK      0x00080000      /* Enable for Rear speakers. */
451						/* Audio output control
452						 * AC97 output enable [5:0]
453						 * I2S output enable [19:16]
454						 * SPDIF output enable [27:24]
455						 */ 
456#define UNKNOWN73               0x73            /* Unknown. Readonly. Default 0x0 */
457#define CHIP_VERSION            0x74            /* P17 Chip version. Channel_id 0 only. Default 00000071 */
458#define EXTENDED_INT_MASK       0x75            /* Used by both playback and capture interrupt handler */
459						/* Sets which Interrupts are enabled. */
460						/* 0x00000001 = Half period. Playback.
461						 * 0x00000010 = Full period. Playback.
462						 * 0x00000100 = Half buffer. Playback.
463						 * 0x00001000 = Full buffer. Playback.
464						 * 0x00010000 = Half buffer. Capture.
465						 * 0x00100000 = Full buffer. Capture.
466						 * Capture can only do 2 periods.
467						 * 0x01000000 = End audio. Playback.
468						 * 0x40000000 = Half buffer Playback,Caputre xrun.
469						 * 0x80000000 = Full buffer Playback,Caputre xrun.
470						 */
471#define EXTENDED_INT            0x76            /* Used by both playback and capture interrupt handler */
472						/* Shows which interrupts are active at the moment. */
473						/* Same bit layout as EXTENDED_INT_MASK */
474#define COUNTER77               0x77		/* Counter range 0 to 0x3fffff, 192000 counts per second. */
475#define COUNTER78               0x78		/* Counter range 0 to 0x3fffff, 44100 counts per second. */
476#define EXTENDED_INT_TIMER      0x79            /* Channel_id 0 only. Used by both playback and capture interrupt handler */
477						/* Causes interrupts based on timer intervals. */
478#define SPI			0x7a		/* SPI: Serial Interface Register */
479#define I2C_A			0x7b		/* I2C Address. 32 bit */
480#define I2C_D0			0x7c		/* I2C Data Port 0. 32 bit */
481#define I2C_D1			0x7d		/* I2C Data Port 1. 32 bit */
482//I2C values
483#define I2C_A_ADC_ADD_MASK	0x000000fe	//The address is a 7 bit address
484#define I2C_A_ADC_RW_MASK	0x00000001	//bit mask for R/W
485#define I2C_A_ADC_TRANS_MASK	0x00000010  	//Bit mask for I2c address DAC value
486#define I2C_A_ADC_ABORT_MASK	0x00000020	//Bit mask for I2C transaction abort flag
487#define I2C_A_ADC_LAST_MASK	0x00000040	//Bit mask for Last word transaction
488#define I2C_A_ADC_BYTE_MASK	0x00000080	//Bit mask for Byte Mode
489
490#define I2C_A_ADC_ADD		0x00000034	//This is the Device address for ADC 
491#define I2C_A_ADC_READ		0x00000001	//To perform a read operation
492#define I2C_A_ADC_START		0x00000100	//Start I2C transaction
493#define I2C_A_ADC_ABORT		0x00000200	//I2C transaction abort
494#define I2C_A_ADC_LAST		0x00000400	//I2C last transaction
495#define I2C_A_ADC_BYTE		0x00000800	//I2C one byte mode
496
497#define I2C_D_ADC_REG_MASK	0xfe000000  	//ADC address register 
498#define I2C_D_ADC_DAT_MASK	0x01ff0000  	//ADC data register
499
500#define ADC_TIMEOUT		0x00000007	//ADC Timeout Clock Disable
501#define ADC_IFC_CTRL		0x0000000b	//ADC Interface Control
502#define ADC_MASTER		0x0000000c	//ADC Master Mode Control
503#define ADC_POWER		0x0000000d	//ADC PowerDown Control
504#define ADC_ATTEN_ADCL		0x0000000e	//ADC Attenuation ADCL
505#define ADC_ATTEN_ADCR		0x0000000f	//ADC Attenuation ADCR
506#define ADC_ALC_CTRL1		0x00000010	//ADC ALC Control 1
507#define ADC_ALC_CTRL2		0x00000011	//ADC ALC Control 2
508#define ADC_ALC_CTRL3		0x00000012	//ADC ALC Control 3
509#define ADC_NOISE_CTRL		0x00000013	//ADC Noise Gate Control
510#define ADC_LIMIT_CTRL		0x00000014	//ADC Limiter Control
511#define ADC_MUX			0x00000015  	//ADC Mux offset
512
513#if 0
514/* FIXME: Not tested yet. */
515#define ADC_GAIN_MASK		0x000000ff	//Mask for ADC Gain
516#define ADC_ZERODB		0x000000cf	//Value to set ADC to 0dB
517#define ADC_MUTE_MASK		0x000000c0	//Mask for ADC mute
518#define ADC_MUTE		0x000000c0	//Value to mute ADC
519#define ADC_OSR			0x00000008	//Mask for ADC oversample rate select
520#define ADC_TIMEOUT_DISABLE	0x00000008	//Value and mask to disable Timeout clock
521#define ADC_HPF_DISABLE		0x00000100	//Value and mask to disable High pass filter
522#define ADC_TRANWIN_MASK	0x00000070	//Mask for Length of Transient Window
523#endif
524
525#define ADC_MUX_MASK		0x0000000f	//Mask for ADC Mux
526#define ADC_MUX_PHONE		0x00000001	//Value to select TAD at ADC Mux (Not used)
527#define ADC_MUX_MIC		0x00000002	//Value to select Mic at ADC Mux
528#define ADC_MUX_LINEIN		0x00000004	//Value to select LineIn at ADC Mux
529#define ADC_MUX_AUX		0x00000008	//Value to select Aux at ADC Mux
530
531#define SET_CHANNEL 0  /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
532#define PCM_FRONT_CHANNEL 0
533#define PCM_REAR_CHANNEL 1
534#define PCM_CENTER_LFE_CHANNEL 2
535#define PCM_UNKNOWN_CHANNEL 3
536#define CONTROL_FRONT_CHANNEL 0
537#define CONTROL_REAR_CHANNEL 3
538#define CONTROL_CENTER_LFE_CHANNEL 1
539#define CONTROL_UNKNOWN_CHANNEL 2
540
541
542/* Based on WM8768 Datasheet Rev 4.2 page 32 */
543#define SPI_REG_MASK	0x1ff	/* 16-bit SPI writes have a 7-bit address */
544#define SPI_REG_SHIFT	9	/* followed by 9 bits of data */
545
546#define SPI_LDA1_REG		0	/* digital attenuation */
547#define SPI_RDA1_REG		1
548#define SPI_LDA2_REG		4
549#define SPI_RDA2_REG		5
550#define SPI_LDA3_REG		6
551#define SPI_RDA3_REG		7
552#define SPI_LDA4_REG		13
553#define SPI_RDA4_REG		14
554#define SPI_MASTDA_REG		8
555
556#define SPI_DA_BIT_UPDATE	(1<<8)	/* update attenuation values */
557#define SPI_DA_BIT_0dB		0xff	/* 0 dB */
558#define SPI_DA_BIT_infdB	0x00	/* inf dB attenuation (mute) */
559
560#define SPI_PL_REG		2
561#define SPI_PL_BIT_L_M		(0<<5)	/* left channel = mute */
562#define SPI_PL_BIT_L_L		(1<<5)	/* left channel = left */
563#define SPI_PL_BIT_L_R		(2<<5)	/* left channel = right */
564#define SPI_PL_BIT_L_C		(3<<5)	/* left channel = (L+R)/2 */
565#define SPI_PL_BIT_R_M		(0<<7)	/* right channel = mute */
566#define SPI_PL_BIT_R_L		(1<<7)	/* right channel = left */
567#define SPI_PL_BIT_R_R		(2<<7)	/* right channel = right */
568#define SPI_PL_BIT_R_C		(3<<7)	/* right channel = (L+R)/2 */
569#define SPI_IZD_REG		2
570#define SPI_IZD_BIT		(0<<4)	/* infinite zero detect */
571
572#define SPI_FMT_REG		3
573#define SPI_FMT_BIT_RJ		(0<<0)	/* right justified mode */
574#define SPI_FMT_BIT_LJ		(1<<0)	/* left justified mode */
575#define SPI_FMT_BIT_I2S		(2<<0)	/* I2S mode */
576#define SPI_FMT_BIT_DSP		(3<<0)	/* DSP Modes A or B */
577#define SPI_LRP_REG		3
578#define SPI_LRP_BIT		(1<<2)	/* invert LRCLK polarity */
579#define SPI_BCP_REG		3
580#define SPI_BCP_BIT		(1<<3)	/* invert BCLK polarity */
581#define SPI_IWL_REG		3
582#define SPI_IWL_BIT_16		(0<<4)	/* 16-bit world length */
583#define SPI_IWL_BIT_20		(1<<4)	/* 20-bit world length */
584#define SPI_IWL_BIT_24		(2<<4)	/* 24-bit world length */
585#define SPI_IWL_BIT_32		(3<<4)	/* 32-bit world length */
586
587#define SPI_MS_REG		10
588#define SPI_MS_BIT		(1<<5)	/* master mode */
589#define SPI_RATE_REG		10	/* only applies in master mode */
590#define SPI_RATE_BIT_128	(0<<6)	/* MCLK = LRCLK * 128 */
591#define SPI_RATE_BIT_192	(1<<6)
592#define SPI_RATE_BIT_256	(2<<6)
593#define SPI_RATE_BIT_384	(3<<6)
594#define SPI_RATE_BIT_512	(4<<6)
595#define SPI_RATE_BIT_768	(5<<6)
596
597/* They really do label the bit for the 4th channel "4" and not "3" */
598#define SPI_DMUTE0_REG		9
599#define SPI_DMUTE1_REG		9
600#define SPI_DMUTE2_REG		9
601#define SPI_DMUTE4_REG		15
602#define SPI_DMUTE0_BIT		(1<<3)
603#define SPI_DMUTE1_BIT		(1<<4)
604#define SPI_DMUTE2_BIT		(1<<5)
605#define SPI_DMUTE4_BIT		(1<<2)
606
607#define SPI_PHASE0_REG		3
608#define SPI_PHASE1_REG		3
609#define SPI_PHASE2_REG		3
610#define SPI_PHASE4_REG		15
611#define SPI_PHASE0_BIT		(1<<6)
612#define SPI_PHASE1_BIT		(1<<7)
613#define SPI_PHASE2_BIT		(1<<8)
614#define SPI_PHASE4_BIT		(1<<3)
615
616#define SPI_PDWN_REG		2	/* power down all DACs */
617#define SPI_PDWN_BIT		(1<<2)
618#define SPI_DACD0_REG		10	/* power down individual DACs */
619#define SPI_DACD1_REG		10
620#define SPI_DACD2_REG		10
621#define SPI_DACD4_REG		15
622#define SPI_DACD0_BIT		(1<<1)
623#define SPI_DACD1_BIT		(1<<2)
624#define SPI_DACD2_BIT		(1<<3)
625#define SPI_DACD4_BIT		(1<<0)	/* datasheet error says it's 1 */
626
627#define SPI_PWRDNALL_REG	10	/* power down everything */
628#define SPI_PWRDNALL_BIT	(1<<4)
629
630#include "ca_midi.h"
631
632struct snd_ca0106;
633
634struct snd_ca0106_channel {
635	struct snd_ca0106 *emu;
636	int number;
637	int use;
638	void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
639	struct snd_ca0106_pcm *epcm;
640};
641
642struct snd_ca0106_pcm {
643	struct snd_ca0106 *emu;
644	struct snd_pcm_substream *substream;
645        int channel_id;
646	unsigned short running;
647};
648
649struct snd_ca0106_details {
650        u32 serial;
651        char * name;
652	int ac97;	/* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
653			   ac97 = 1 -> Default to AC97 in. */
654	int gpio_type;	/* gpio_type = 1 -> shared mic-in/line-in
655			   gpio_type = 2 -> shared side-out/line-in. */
656	int i2c_adc;	/* with i2c_adc=1, the driver adds some capture volume
657			   controls, phone, mic, line-in and aux. */
658	u16 spi_dac;	/* spi_dac = 0 -> no spi interface for DACs
659			   spi_dac = 0x<front><rear><center-lfe><side>
660			   -> specifies DAC id for each channel pair. */
661};
662
663// definition of the chip-specific record
664struct snd_ca0106 {
665	struct snd_card *card;
666	const struct snd_ca0106_details *details;
667	struct pci_dev *pci;
668
669	unsigned long port;
670	struct resource *res_port;
671	int irq;
672
673	unsigned int serial;            /* serial number */
674	unsigned short model;		/* subsystem id */
675
676	spinlock_t emu_lock;
677
678	struct snd_ac97 *ac97;
679	struct snd_pcm *pcm[4];
680
681	struct snd_ca0106_channel playback_channels[4];
682	struct snd_ca0106_channel capture_channels[4];
683	u32 spdif_bits[4];             /* s/pdif out default setup */
684	u32 spdif_str_bits[4];         /* s/pdif out per-stream setup */
685	int spdif_enable;
686	int capture_source;
687	int i2c_capture_source;
688	u8 i2c_capture_volume[4][2];
689	int capture_mic_line_in;
690
691	struct snd_dma_buffer buffer;
692
693	struct snd_ca_midi midi;
694	struct snd_ca_midi midi2;
695
696	u16 spi_dac_reg[16];
697
698#ifdef CONFIG_PM_SLEEP
699#define NUM_SAVED_VOLUMES	9
700	unsigned int saved_vol[NUM_SAVED_VOLUMES];
701#endif
702};
703
704int snd_ca0106_mixer(struct snd_ca0106 *emu);
705int snd_ca0106_proc_init(struct snd_ca0106 * emu);
706
707unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, 
708				 unsigned int reg, 
709				 unsigned int chn);
710
711void snd_ca0106_ptr_write(struct snd_ca0106 *emu, 
712			  unsigned int reg, 
713			  unsigned int chn, 
714			  unsigned int data);
715
716int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
717
718int snd_ca0106_spi_write(struct snd_ca0106 * emu,
719				   unsigned int data);
720
721#ifdef CONFIG_PM_SLEEP
722void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
723void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
724#else
725#define snd_ca0106_mixer_suspend(chip)	do { } while (0)
726#define snd_ca0106_mixer_resume(chip)	do { } while (0)
727#endif
v3.15
 
  1/*
  2 *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
  3 *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
  4 *  Version: 0.0.22
  5 *
  6 *  FEATURES currently supported:
  7 *    See ca0106_main.c for features.
  8 * 
  9 *  Changelog:
 10 *    Support interrupts per period.
 11 *    Removed noise from Center/LFE channel when in Analog mode.
 12 *    Rename and remove mixer controls.
 13 *  0.0.6
 14 *    Use separate card based DMA buffer for periods table list.
 15 *  0.0.7
 16 *    Change remove and rename ctrls into lists.
 17 *  0.0.8
 18 *    Try to fix capture sources.
 19 *  0.0.9
 20 *    Fix AC3 output.
 21 *    Enable S32_LE format support.
 22 *  0.0.10
 23 *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
 24 *  0.0.11
 25 *    Add Model name recognition.
 26 *  0.0.12
 27 *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
 28 *    Remove redundent "voice" handling.
 29 *  0.0.13
 30 *    Single trigger call for multi channels.
 31 *  0.0.14
 32 *    Set limits based on what the sound card hardware can do.
 33 *    playback periods_min=2, periods_max=8
 34 *    capture hw constraints require period_size = n * 64 bytes.
 35 *    playback hw constraints require period_size = n * 64 bytes.
 36 *  0.0.15
 37 *    Separated ca0106.c into separate functional .c files.
 38 *  0.0.16
 39 *    Implement 192000 sample rate.
 40 *  0.0.17
 41 *    Add support for SB0410 and SB0413.
 42 *  0.0.18
 43 *    Modified Copyright message.
 44 *  0.0.19
 45 *    Added I2C and SPI registers. Filled in interrupt enable.
 46 *  0.0.20
 47 *    Added GPIO info for SB Live 24bit.
 48 *  0.0.21
 49 *   Implement support for Line-in capture on SB Live 24bit.
 50 *  0.0.22
 51 *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
 52 *
 53 *
 54 *  This code was initially based on code from ALSA's emu10k1x.c which is:
 55 *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
 56 *
 57 *   This program is free software; you can redistribute it and/or modify
 58 *   it under the terms of the GNU General Public License as published by
 59 *   the Free Software Foundation; either version 2 of the License, or
 60 *   (at your option) any later version.
 61 *
 62 *   This program is distributed in the hope that it will be useful,
 63 *   but WITHOUT ANY WARRANTY; without even the implied warranty of
 64 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 65 *   GNU General Public License for more details.
 66 *
 67 *   You should have received a copy of the GNU General Public License
 68 *   along with this program; if not, write to the Free Software
 69 *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
 70 *
 71 */
 72
 73/************************************************************************************************/
 74/* PCI function 0 registers, address = <val> + PCIBASE0						*/
 75/************************************************************************************************/
 76
 77#define PTR			0x00		/* Indexed register set pointer register	*/
 78						/* NOTE: The CHANNELNUM and ADDRESS words can	*/
 79						/* be modified independently of each other.	*/
 80						/* CNL[1:0], ADDR[27:16]                        */
 81
 82#define DATA			0x04		/* Indexed register set data register		*/
 83						/* DATA[31:0]					*/
 84
 85#define IPR			0x08		/* Global interrupt pending register		*/
 86						/* Clear pending interrupts by writing a 1 to	*/
 87						/* the relevant bits and zero to the other bits	*/
 88#define IPR_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
 89#define IPR_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
 90#define IPR_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
 91#define IPR_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
 92#define IPR_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
 93#define IPR_SPI			0x00000800      /* SPI transaction completed			*/
 94#define IPR_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
 95#define IPR_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
 96#define IPR_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x76	*/
 97#define IPR_GPI			0x00000080      /* General Purpose input changed		*/
 98#define IPR_SRC_LOCKED          0x00000040      /* SRC lock status changed			*/
 99#define IPR_SPDIF_STATUS        0x00000020      /* SPDIF status changed				*/
100#define IPR_TIMER2              0x00000010      /* 192000Hz Timer				*/
101#define IPR_TIMER1              0x00000008      /* 44100Hz Timer				*/
102#define IPR_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
103#define IPR_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
104#define IPR_PCI			0x00000001	/* PCI Bus error				*/
105
106#define INTE			0x0c		/* Interrupt enable register			*/
107
108#define INTE_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
109#define INTE_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
110#define INTE_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
111#define INTE_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
112#define INTE_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
113#define INTE_SPI		0x00000800      /* SPI transaction completed			*/
114#define INTE_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
115#define INTE_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
116#define INTE_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x75 */
117#define INTE_GPI		0x00000080      /* General Purpose input changed		*/
118#define INTE_SRC_LOCKED         0x00000040      /* SRC lock status changed			*/
119#define INTE_SPDIF_STATUS       0x00000020      /* SPDIF status changed				*/
120#define INTE_TIMER2             0x00000010      /* 192000Hz Timer				*/
121#define INTE_TIMER1             0x00000008      /* 44100Hz Timer				*/
122#define INTE_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
123#define INTE_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
124#define INTE_PCI		0x00000001	/* PCI Bus error				*/
125
126#define UNKNOWN10		0x10		/* Unknown ??. Defaults to 0 */
127#define HCFG			0x14		/* Hardware config register			*/
128						/* 0x1000 causes AC3 to fails. It adds a dither bit. */
129
130#define HCFG_STAC		0x10000000	/* Special mode for STAC9460 Codec. */
131#define HCFG_CAPTURE_I2S_BYPASS	0x08000000	/* 1 = bypass I2S input async SRC. */
132#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000	/* 1 = bypass SPDIF input async SRC. */
133#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000	/* 0 = I2S IN mixer output, 1 = I2S IN1. */
134#define HCFG_FORCE_LOCK		0x01000000	/* For test only. Force input SRC tracker to lock. */
135#define HCFG_PLAYBACK_ATTENUATION 0x00006000	/* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
136#define HCFG_PLAYBACK_DITHER	0x00001000	/* 1 = Add dither bit to all playback channels. */
137#define HCFG_PLAYBACK_S32_LE	0x00000800	/* 1 = S32_LE, 0 = S16_LE                       */
138#define HCFG_CAPTURE_S32_LE	0x00000400	/* 1 = S32_LE, 0 = S16_LE (S32_LE current not working)	*/
139#define HCFG_8_CHANNEL_PLAY	0x00000200	/* 1 = 8 channels, 0 = 2 channels per substream.*/
140#define HCFG_8_CHANNEL_CAPTURE	0x00000100	/* 1 = 8 channels, 0 = 2 channels per substream.*/
141#define HCFG_MONO		0x00000080	/* 1 = I2S Input mono                           */
142#define HCFG_I2S_OUTPUT		0x00000010	/* 1 = I2S Output disabled                      */
143#define HCFG_AC97		0x00000008	/* 0 = AC97 1.0, 1 = AC97 2.0                   */
144#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004	/* 1 = Cancel bustmaster accesses to soundcache */
145						/* NOTE: This should generally never be used.  	*/
146#define HCFG_LOCK_CAPTURE_CACHE	0x00000002	/* 1 = Cancel bustmaster accesses to soundcache */
147						/* NOTE: This should generally never be used.  	*/
148#define HCFG_AUDIOENABLE	0x00000001	/* 0 = CODECs transmit zero-valued samples	*/
149						/* Should be set to 1 when the EMU10K1 is	*/
150						/* completely initialized.			*/
151#define GPIO			0x18		/* Defaults: 005f03a3-Analog, 005f02a2-SPDIF.   */
152						/* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
153						/* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
154						/* SB Live 24bit:
155						 * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
156						 * bit 9 0 = Mute / 1 = Analog out.
157						 * bit 10 0 = Line-in / 1 = Mic-in.
158						 * bit 11 0 = ? / 1 = ?
159						 * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
160						 * bit 13 0 = ? / 1 = ?
161						 * bit 14 0 = Mute / 1 = Analog out
162						 * bit 15 0 = ? / 1 = ?
163						 * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
164						 */
165						/* 8 general purpose programmable In/Out pins.
166						 * GPI [8:0] Read only. Default 0.
167						 * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
168						 * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
169						 */
170#define AC97DATA		0x1c		/* AC97 register set data register (16 bit)	*/
171
172#define AC97ADDRESS		0x1e		/* AC97 register set address register (8 bit)	*/
173
174/********************************************************************************************************/
175/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
176/********************************************************************************************************/
177                                                                                                                           
178/* Initially all registers from 0x00 to 0x3f have zero contents. */
179#define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
180						/* One list entry: 4 bytes for DMA address, 
181						 * 4 bytes for period_size << 16.
182						 * One list entry is 8 bytes long.
183						 * One list entry for each period in the buffer.
184						 */
185						/* ADDR[31:0], Default: 0x0 */
186#define PLAYBACK_LIST_SIZE	0x01		/* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000  */
187						/* SIZE[21:16], Default: 0x8 */
188#define PLAYBACK_LIST_PTR	0x02		/* Pointer to the current period being played */
189						/* PTR[5:0], Default: 0x0 */
190#define PLAYBACK_UNKNOWN3	0x03		/* Not used ?? */
191#define PLAYBACK_DMA_ADDR	0x04		/* Playback DMA address */
192						/* DMA[31:0], Default: 0x0 */
193#define PLAYBACK_PERIOD_SIZE	0x05		/* Playback period size. win2000 uses 0x04000000 */
194						/* SIZE[31:16], Default: 0x0 */
195#define PLAYBACK_POINTER	0x06		/* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
196						/* POINTER[15:0], Default: 0x0 */
197#define PLAYBACK_PERIOD_END_ADDR 0x07		/* Playback fifo end address */
198						/* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
199#define PLAYBACK_FIFO_OFFSET_ADDRESS	0x08	/* Current fifo offset address [21:16] */
200						/* Cache size valid [5:0] */
201#define PLAYBACK_UNKNOWN9	0x09		/* 0x9 to 0xf Unused */
202#define CAPTURE_DMA_ADDR	0x10		/* Capture DMA address */
203						/* DMA[31:0], Default: 0x0 */
204#define CAPTURE_BUFFER_SIZE	0x11		/* Capture buffer size */
205						/* SIZE[31:16], Default: 0x0 */
206#define CAPTURE_POINTER		0x12		/* Capture buffer pointer. Sample currently in ADC */
207						/* POINTER[15:0], Default: 0x0 */
208#define CAPTURE_FIFO_OFFSET_ADDRESS	0x13	/* Current fifo offset address [21:16] */
209						/* Cache size valid [5:0] */
210#define PLAYBACK_LAST_SAMPLE    0x20		/* The sample currently being played */
211/* 0x21 - 0x3f unused */
212#define BASIC_INTERRUPT         0x40		/* Used by both playback and capture interrupt handler */
213						/* Playback (0x1<<channel_id) */
214						/* Capture  (0x100<<channel_id) */
215						/* Playback sample rate 96000 = 0x20000 */
216						/* Start Playback [3:0] (one bit per channel)
217						 * Start Capture [11:8] (one bit per channel)
218						 * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
219						 * Playback mixer in enable [27:24] (one bit per channel)
220						 * Playback mixer out enable [31:28] (one bit per channel)
221						 */
222/* The Digital out jack is shared with the Center/LFE Analogue output. 
223 * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
224 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
225 * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
226 * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
227 * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
228 */
229/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
230 * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
231 * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
232 * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
233 */
234/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
235 * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. 
236 */
237#define SPCS0			0x41		/* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006	*/
238#define SPCS1			0x42		/* SPDIF output Channel Status 1 register. For Front */
239#define SPCS2			0x43		/* SPDIF output Channel Status 2 register. For Center/LFE */
240#define SPCS3			0x44		/* SPDIF output Channel Status 3 register. Unknown */
241						/* When Channel set to 0: */
242#define SPCS_CLKACCYMASK	0x30000000	/* Clock accuracy				*/
243#define SPCS_CLKACCY_1000PPM	0x00000000	/* 1000 parts per million			*/
244#define SPCS_CLKACCY_50PPM	0x10000000	/* 50 parts per million				*/
245#define SPCS_CLKACCY_VARIABLE	0x20000000	/* Variable accuracy				*/
246#define SPCS_SAMPLERATEMASK	0x0f000000	/* Sample rate					*/
247#define SPCS_SAMPLERATE_44	0x00000000	/* 44.1kHz sample rate				*/
248#define SPCS_SAMPLERATE_48	0x02000000	/* 48kHz sample rate				*/
249#define SPCS_SAMPLERATE_32	0x03000000	/* 32kHz sample rate				*/
250#define SPCS_CHANNELNUMMASK	0x00f00000	/* Channel number				*/
251#define SPCS_CHANNELNUM_UNSPEC	0x00000000	/* Unspecified channel number			*/
252#define SPCS_CHANNELNUM_LEFT	0x00100000	/* Left channel					*/
253#define SPCS_CHANNELNUM_RIGHT	0x00200000	/* Right channel				*/
254#define SPCS_SOURCENUMMASK	0x000f0000	/* Source number				*/
255#define SPCS_SOURCENUM_UNSPEC	0x00000000	/* Unspecified source number			*/
256#define SPCS_GENERATIONSTATUS	0x00008000	/* Originality flag (see IEC-958 spec)		*/
257#define SPCS_CATEGORYCODEMASK	0x00007f00	/* Category code (see IEC-958 spec)		*/
258#define SPCS_MODEMASK		0x000000c0	/* Mode (see IEC-958 spec)			*/
259#define SPCS_EMPHASISMASK	0x00000038	/* Emphasis					*/
260#define SPCS_EMPHASIS_NONE	0x00000000	/* No emphasis					*/
261#define SPCS_EMPHASIS_50_15	0x00000008	/* 50/15 usec 2 channel				*/
262#define SPCS_COPYRIGHT		0x00000004	/* Copyright asserted flag -- do not modify	*/
263#define SPCS_NOTAUDIODATA	0x00000002	/* 0 = Digital audio, 1 = not audio		*/
264#define SPCS_PROFESSIONAL	0x00000001	/* 0 = Consumer (IEC-958), 1 = pro (AES3-1992)	*/
265
266						/* When Channel set to 1: */
267#define SPCS_WORD_LENGTH_MASK	0x0000000f	/* Word Length Mask				*/
268#define SPCS_WORD_LENGTH_16	0x00000008	/* Word Length 16 bit				*/
269#define SPCS_WORD_LENGTH_17	0x00000006	/* Word Length 17 bit				*/
270#define SPCS_WORD_LENGTH_18	0x00000004	/* Word Length 18 bit				*/
271#define SPCS_WORD_LENGTH_19	0x00000002	/* Word Length 19 bit				*/
272#define SPCS_WORD_LENGTH_20A	0x0000000a	/* Word Length 20 bit				*/
273#define SPCS_WORD_LENGTH_20	0x00000009	/* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
274#define SPCS_WORD_LENGTH_21	0x00000007	/* Word Length 21 bit				*/
275#define SPCS_WORD_LENGTH_22	0x00000005	/* Word Length 22 bit				*/
276#define SPCS_WORD_LENGTH_23	0x00000003	/* Word Length 23 bit				*/
277#define SPCS_WORD_LENGTH_24	0x0000000b	/* Word Length 24 bit				*/
278#define SPCS_ORIGINAL_SAMPLE_RATE_MASK	0x000000f0 /* Original Sample rate			*/
279#define SPCS_ORIGINAL_SAMPLE_RATE_NONE	0x00000000 /* Original Sample rate not indicated	*/
280#define SPCS_ORIGINAL_SAMPLE_RATE_16000	0x00000010 /* Original Sample rate	*/
281#define SPCS_ORIGINAL_SAMPLE_RATE_RES1	0x00000020 /* Original Sample rate	*/
282#define SPCS_ORIGINAL_SAMPLE_RATE_32000	0x00000030 /* Original Sample rate	*/
283#define SPCS_ORIGINAL_SAMPLE_RATE_12000	0x00000040 /* Original Sample rate	*/
284#define SPCS_ORIGINAL_SAMPLE_RATE_11025	0x00000050 /* Original Sample rate	*/
285#define SPCS_ORIGINAL_SAMPLE_RATE_8000	0x00000060 /* Original Sample rate	*/
286#define SPCS_ORIGINAL_SAMPLE_RATE_RES2	0x00000070 /* Original Sample rate	*/
287#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate	*/
288#define SPCS_ORIGINAL_SAMPLE_RATE_24000	0x00000090 /* Original Sample rate	*/
289#define SPCS_ORIGINAL_SAMPLE_RATE_96000	0x000000a0 /* Original Sample rate	*/
290#define SPCS_ORIGINAL_SAMPLE_RATE_48000	0x000000b0 /* Original Sample rate	*/
291#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate	*/
292#define SPCS_ORIGINAL_SAMPLE_RATE_22050	0x000000d0 /* Original Sample rate	*/
293#define SPCS_ORIGINAL_SAMPLE_RATE_88200	0x000000e0 /* Original Sample rate	*/
294#define SPCS_ORIGINAL_SAMPLE_RATE_44100	0x000000f0 /* Original Sample rate	*/
295
296#define SPDIF_SELECT1		0x45		/* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
297						/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
298						 * But as the jack is shared, use 0xf00.
299						 * The Windows2000 driver uses 0x0000000f for both digital and analog.
300						 * 0xf00 introduces interesting noises onto the Center/LFE.
301						 * If you turn the volume up, you hear computer noise,
302						 * e.g. mouse moving, changing between app windows etc.
303						 * So, I am going to set this to 0x0000000f all the time now,
304						 * same as the windows driver does.
305						 * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
306						 */
307						/* When Channel = 0:
308						 * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
309						 * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
310						 * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
311						 */
312						/* When Channel = 1:
313						 * SPDIF 0 User data [7:0]
314						 * SPDIF 1 User data [15:8]
315						 * SPDIF 0 User data [23:16]
316						 * SPDIF 0 User data [31:24]
317						 * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
318						 */
319#define WATERMARK		0x46		/* Test bit to indicate cache usage level */
320#define SPDIF_INPUT_STATUS	0x49		/* SPDIF Input status register. Bits the same as SPCS.
321						 * When Channel = 0: Bits the same as SPCS channel 0.
322						 * When Channel = 1: Bits the same as SPCS channel 1.
323						 * When Channel = 2:
324						 * SPDIF Input User data [16:0]
325						 * SPDIF Input Frame count [21:16]
326						 */
327#define CAPTURE_CACHE_DATA	0x50		/* 0x50-0x5f Recorded samples. */
328#define CAPTURE_SOURCE          0x60            /* Capture Source 0 = MIC */
329#define CAPTURE_SOURCE_CHANNEL0 0xf0000000	/* Mask for selecting the Capture sources */
330#define CAPTURE_SOURCE_CHANNEL1 0x0f000000	/* 0 - SPDIF mixer output. */
331#define CAPTURE_SOURCE_CHANNEL2 0x00f00000      /* 1 - What you hear or . 2 - ?? */
332#define CAPTURE_SOURCE_CHANNEL3 0x000f0000	/* 3 - Mic in, Line in, TAD in, Aux in. */
333#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff	/* Default 0x00e4 */
334						/* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 
335						 * Record source select for channel 0 [18:16]
336						 * Record source select for channel 1 [22:20]
337						 * Record source select for channel 2 [26:24]
338						 * Record source select for channel 3 [30:28]
339						 * 0 - SPDIF mixer output.
340						 * 1 - i2s mixer output.
341						 * 2 - SPDIF input.
342						 * 3 - i2s input.
343						 * 4 - AC97 capture.
344						 * 5 - SRC output.
345						 */
346#define CAPTURE_VOLUME1         0x61            /* Capture  volume per channel 0-3 */
347#define CAPTURE_VOLUME2         0x62            /* Capture  volume per channel 4-7 */
348
349#define PLAYBACK_ROUTING1       0x63            /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
350#define ROUTING1_REAR           0x77000000      /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
351#define ROUTING1_NULL           0x00770000      /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
352#define ROUTING1_CENTER_LFE     0x00007700      /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
353#define ROUTING1_FRONT          0x00000077	/* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
354						/* Channel_id's handle stereo channels. Channel X is a single mono channel */
355						/* Host is input from the PCI bus. */
356						/* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
357						 * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
358						 * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
359						 * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
360						 * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
361						 * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
362						 * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
363						 * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
364						 */
365
366#define PLAYBACK_ROUTING2       0x64            /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
367						/* SRC is input from the capture inputs. */
368						/* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
369						 * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
370						 * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
371						 * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
372						 * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
373						 * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
374						 * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
375						 * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
376						 */
377
378#define PLAYBACK_MUTE           0x65            /* Unknown. While playing 0x0, while silent 0x00fc0000 */
379						/* SPDIF Mixer input control:
380						 * Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
381						 * Invert Host to SPDIF Mixer [15:8] (One bit per channel)
382						 * SRC to SPDIF Mixer disable [23:16] (One bit per channel)
383						 * Host to SPDIF Mixer disable [31:24] (One bit per channel)
384						 */
385#define PLAYBACK_VOLUME1        0x66            /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
386						/* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
387						/* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
388						/* One register for each of the 4 stereo streams. */
389						/* SRC Right volume [7:0]
390						 * SRC Left  volume [15:8]
391						 * Host Right volume [23:16]
392						 * Host Left  volume [31:24]
393						 */
394#define CAPTURE_ROUTING1        0x67            /* Capture Routing. Default 0x32765410 */
395						/* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
396#define CAPTURE_ROUTING2        0x68            /* Unknown Routing. Default 0x76767676 */
397						/* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
398#define CAPTURE_MUTE            0x69            /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
399						/* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
400#define PLAYBACK_VOLUME2        0x6a            /* Playback Analog volume per channel. Does not effect AC3 output */
401						/* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
402#define UNKNOWN6b               0x6b            /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
403#define MIDI_UART_A_DATA		0x6c            /* Midi Uart A Data */
404#define MIDI_UART_A_CMD		0x6d            /* Midi Uart A Command/Status */
405#define MIDI_UART_B_DATA		0x6e            /* Midi Uart B Data (currently unused) */
406#define MIDI_UART_B_CMD		0x6f            /* Midi Uart B Command/Status (currently unused) */
407
408/* unique channel identifier for midi->channel */
409
410#define CA0106_MIDI_CHAN_A		0x1
411#define CA0106_MIDI_CHAN_B		0x2
412
413/* from mpu401 */
414
415#define CA0106_MIDI_INPUT_AVAIL 	0x80
416#define CA0106_MIDI_OUTPUT_READY	0x40
417#define CA0106_MPU401_RESET		0xff
418#define CA0106_MPU401_ENTER_UART	0x3f
419#define CA0106_MPU401_ACK		0xfe
420
421#define SAMPLE_RATE_TRACKER_STATUS 0x70         /* Readonly. Default 00108000 00108000 00500000 00500000 */
422						/* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 =  1.0
423						 * Rate Locked [20]
424						 * SPDIF Locked [21] For SPDIF channel only.
425						 * Valid Audio [22] For SPDIF channel only.
426						 */
427#define CAPTURE_CONTROL         0x71            /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
428						/* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
429						/* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
430						/* Sample rate output control register Channel=0
431						 * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
432						 * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
433						 * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
434						 * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
435						 * Record mixer output enable [12:10] 
436						 * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
437						 * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
438						 * I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
439						 * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
440						 * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
441						 * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
442						 * I2S input mode [23] (0=Slave, 1=Master)
443						 * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
444						 * SPDIF output source select [26] (0=host, 1=SRC)
445						 * Not used [27]
446						 * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
447						 * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
448						 */ 
449						/* Sample rate output control register Channel=1
450						 * I2S Input 0 volume Right [7:0]
451						 * I2S Input 0 volume Left [15:8]
452						 * I2S Input 1 volume Right [23:16]
453						 * I2S Input 1 volume Left [31:24]
454						 */
455						/* Sample rate output control register Channel=2
456						 * SPDIF Input volume Right [23:16]
457						 * SPDIF Input volume Left [31:24]
458						 */
459						/* Sample rate output control register Channel=3
460						 * No used
461						 */
462#define SPDIF_SELECT2           0x72            /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
463#define ROUTING2_FRONT_MASK     0x00010000      /* Enable for Front speakers. */
464#define ROUTING2_CENTER_LFE_MASK 0x00020000     /* Enable for Center/LFE speakers. */
465#define ROUTING2_REAR_MASK      0x00080000      /* Enable for Rear speakers. */
466						/* Audio output control
467						 * AC97 output enable [5:0]
468						 * I2S output enable [19:16]
469						 * SPDIF output enable [27:24]
470						 */ 
471#define UNKNOWN73               0x73            /* Unknown. Readonly. Default 0x0 */
472#define CHIP_VERSION            0x74            /* P17 Chip version. Channel_id 0 only. Default 00000071 */
473#define EXTENDED_INT_MASK       0x75            /* Used by both playback and capture interrupt handler */
474						/* Sets which Interrupts are enabled. */
475						/* 0x00000001 = Half period. Playback.
476						 * 0x00000010 = Full period. Playback.
477						 * 0x00000100 = Half buffer. Playback.
478						 * 0x00001000 = Full buffer. Playback.
479						 * 0x00010000 = Half buffer. Capture.
480						 * 0x00100000 = Full buffer. Capture.
481						 * Capture can only do 2 periods.
482						 * 0x01000000 = End audio. Playback.
483						 * 0x40000000 = Half buffer Playback,Caputre xrun.
484						 * 0x80000000 = Full buffer Playback,Caputre xrun.
485						 */
486#define EXTENDED_INT            0x76            /* Used by both playback and capture interrupt handler */
487						/* Shows which interrupts are active at the moment. */
488						/* Same bit layout as EXTENDED_INT_MASK */
489#define COUNTER77               0x77		/* Counter range 0 to 0x3fffff, 192000 counts per second. */
490#define COUNTER78               0x78		/* Counter range 0 to 0x3fffff, 44100 counts per second. */
491#define EXTENDED_INT_TIMER      0x79            /* Channel_id 0 only. Used by both playback and capture interrupt handler */
492						/* Causes interrupts based on timer intervals. */
493#define SPI			0x7a		/* SPI: Serial Interface Register */
494#define I2C_A			0x7b		/* I2C Address. 32 bit */
495#define I2C_D0			0x7c		/* I2C Data Port 0. 32 bit */
496#define I2C_D1			0x7d		/* I2C Data Port 1. 32 bit */
497//I2C values
498#define I2C_A_ADC_ADD_MASK	0x000000fe	//The address is a 7 bit address
499#define I2C_A_ADC_RW_MASK	0x00000001	//bit mask for R/W
500#define I2C_A_ADC_TRANS_MASK	0x00000010  	//Bit mask for I2c address DAC value
501#define I2C_A_ADC_ABORT_MASK	0x00000020	//Bit mask for I2C transaction abort flag
502#define I2C_A_ADC_LAST_MASK	0x00000040	//Bit mask for Last word transaction
503#define I2C_A_ADC_BYTE_MASK	0x00000080	//Bit mask for Byte Mode
504
505#define I2C_A_ADC_ADD		0x00000034	//This is the Device address for ADC 
506#define I2C_A_ADC_READ		0x00000001	//To perform a read operation
507#define I2C_A_ADC_START		0x00000100	//Start I2C transaction
508#define I2C_A_ADC_ABORT		0x00000200	//I2C transaction abort
509#define I2C_A_ADC_LAST		0x00000400	//I2C last transaction
510#define I2C_A_ADC_BYTE		0x00000800	//I2C one byte mode
511
512#define I2C_D_ADC_REG_MASK	0xfe000000  	//ADC address register 
513#define I2C_D_ADC_DAT_MASK	0x01ff0000  	//ADC data register
514
515#define ADC_TIMEOUT		0x00000007	//ADC Timeout Clock Disable
516#define ADC_IFC_CTRL		0x0000000b	//ADC Interface Control
517#define ADC_MASTER		0x0000000c	//ADC Master Mode Control
518#define ADC_POWER		0x0000000d	//ADC PowerDown Control
519#define ADC_ATTEN_ADCL		0x0000000e	//ADC Attenuation ADCL
520#define ADC_ATTEN_ADCR		0x0000000f	//ADC Attenuation ADCR
521#define ADC_ALC_CTRL1		0x00000010	//ADC ALC Control 1
522#define ADC_ALC_CTRL2		0x00000011	//ADC ALC Control 2
523#define ADC_ALC_CTRL3		0x00000012	//ADC ALC Control 3
524#define ADC_NOISE_CTRL		0x00000013	//ADC Noise Gate Control
525#define ADC_LIMIT_CTRL		0x00000014	//ADC Limiter Control
526#define ADC_MUX			0x00000015  	//ADC Mux offset
527
528#if 0
529/* FIXME: Not tested yet. */
530#define ADC_GAIN_MASK		0x000000ff	//Mask for ADC Gain
531#define ADC_ZERODB		0x000000cf	//Value to set ADC to 0dB
532#define ADC_MUTE_MASK		0x000000c0	//Mask for ADC mute
533#define ADC_MUTE		0x000000c0	//Value to mute ADC
534#define ADC_OSR			0x00000008	//Mask for ADC oversample rate select
535#define ADC_TIMEOUT_DISABLE	0x00000008	//Value and mask to disable Timeout clock
536#define ADC_HPF_DISABLE		0x00000100	//Value and mask to disable High pass filter
537#define ADC_TRANWIN_MASK	0x00000070	//Mask for Length of Transient Window
538#endif
539
540#define ADC_MUX_MASK		0x0000000f	//Mask for ADC Mux
541#define ADC_MUX_PHONE		0x00000001	//Value to select TAD at ADC Mux (Not used)
542#define ADC_MUX_MIC		0x00000002	//Value to select Mic at ADC Mux
543#define ADC_MUX_LINEIN		0x00000004	//Value to select LineIn at ADC Mux
544#define ADC_MUX_AUX		0x00000008	//Value to select Aux at ADC Mux
545
546#define SET_CHANNEL 0  /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
547#define PCM_FRONT_CHANNEL 0
548#define PCM_REAR_CHANNEL 1
549#define PCM_CENTER_LFE_CHANNEL 2
550#define PCM_UNKNOWN_CHANNEL 3
551#define CONTROL_FRONT_CHANNEL 0
552#define CONTROL_REAR_CHANNEL 3
553#define CONTROL_CENTER_LFE_CHANNEL 1
554#define CONTROL_UNKNOWN_CHANNEL 2
555
556
557/* Based on WM8768 Datasheet Rev 4.2 page 32 */
558#define SPI_REG_MASK	0x1ff	/* 16-bit SPI writes have a 7-bit address */
559#define SPI_REG_SHIFT	9	/* followed by 9 bits of data */
560
561#define SPI_LDA1_REG		0	/* digital attenuation */
562#define SPI_RDA1_REG		1
563#define SPI_LDA2_REG		4
564#define SPI_RDA2_REG		5
565#define SPI_LDA3_REG		6
566#define SPI_RDA3_REG		7
567#define SPI_LDA4_REG		13
568#define SPI_RDA4_REG		14
569#define SPI_MASTDA_REG		8
570
571#define SPI_DA_BIT_UPDATE	(1<<8)	/* update attenuation values */
572#define SPI_DA_BIT_0dB		0xff	/* 0 dB */
573#define SPI_DA_BIT_infdB	0x00	/* inf dB attenuation (mute) */
574
575#define SPI_PL_REG		2
576#define SPI_PL_BIT_L_M		(0<<5)	/* left channel = mute */
577#define SPI_PL_BIT_L_L		(1<<5)	/* left channel = left */
578#define SPI_PL_BIT_L_R		(2<<5)	/* left channel = right */
579#define SPI_PL_BIT_L_C		(3<<5)	/* left channel = (L+R)/2 */
580#define SPI_PL_BIT_R_M		(0<<7)	/* right channel = mute */
581#define SPI_PL_BIT_R_L		(1<<7)	/* right channel = left */
582#define SPI_PL_BIT_R_R		(2<<7)	/* right channel = right */
583#define SPI_PL_BIT_R_C		(3<<7)	/* right channel = (L+R)/2 */
584#define SPI_IZD_REG		2
585#define SPI_IZD_BIT		(1<<4)	/* infinite zero detect */
586
587#define SPI_FMT_REG		3
588#define SPI_FMT_BIT_RJ		(0<<0)	/* right justified mode */
589#define SPI_FMT_BIT_LJ		(1<<0)	/* left justified mode */
590#define SPI_FMT_BIT_I2S		(2<<0)	/* I2S mode */
591#define SPI_FMT_BIT_DSP		(3<<0)	/* DSP Modes A or B */
592#define SPI_LRP_REG		3
593#define SPI_LRP_BIT		(1<<2)	/* invert LRCLK polarity */
594#define SPI_BCP_REG		3
595#define SPI_BCP_BIT		(1<<3)	/* invert BCLK polarity */
596#define SPI_IWL_REG		3
597#define SPI_IWL_BIT_16		(0<<4)	/* 16-bit world length */
598#define SPI_IWL_BIT_20		(1<<4)	/* 20-bit world length */
599#define SPI_IWL_BIT_24		(2<<4)	/* 24-bit world length */
600#define SPI_IWL_BIT_32		(3<<4)	/* 32-bit world length */
601
602#define SPI_MS_REG		10
603#define SPI_MS_BIT		(1<<5)	/* master mode */
604#define SPI_RATE_REG		10	/* only applies in master mode */
605#define SPI_RATE_BIT_128	(0<<6)	/* MCLK = LRCLK * 128 */
606#define SPI_RATE_BIT_192	(1<<6)
607#define SPI_RATE_BIT_256	(2<<6)
608#define SPI_RATE_BIT_384	(3<<6)
609#define SPI_RATE_BIT_512	(4<<6)
610#define SPI_RATE_BIT_768	(5<<6)
611
612/* They really do label the bit for the 4th channel "4" and not "3" */
613#define SPI_DMUTE0_REG		9
614#define SPI_DMUTE1_REG		9
615#define SPI_DMUTE2_REG		9
616#define SPI_DMUTE4_REG		15
617#define SPI_DMUTE0_BIT		(1<<3)
618#define SPI_DMUTE1_BIT		(1<<4)
619#define SPI_DMUTE2_BIT		(1<<5)
620#define SPI_DMUTE4_BIT		(1<<2)
621
622#define SPI_PHASE0_REG		3
623#define SPI_PHASE1_REG		3
624#define SPI_PHASE2_REG		3
625#define SPI_PHASE4_REG		15
626#define SPI_PHASE0_BIT		(1<<6)
627#define SPI_PHASE1_BIT		(1<<7)
628#define SPI_PHASE2_BIT		(1<<8)
629#define SPI_PHASE4_BIT		(1<<3)
630
631#define SPI_PDWN_REG		2	/* power down all DACs */
632#define SPI_PDWN_BIT		(1<<2)
633#define SPI_DACD0_REG		10	/* power down individual DACs */
634#define SPI_DACD1_REG		10
635#define SPI_DACD2_REG		10
636#define SPI_DACD4_REG		15
637#define SPI_DACD0_BIT		(1<<1)
638#define SPI_DACD1_BIT		(1<<2)
639#define SPI_DACD2_BIT		(1<<3)
640#define SPI_DACD4_BIT		(1<<0)	/* datasheet error says it's 1 */
641
642#define SPI_PWRDNALL_REG	10	/* power down everything */
643#define SPI_PWRDNALL_BIT	(1<<4)
644
645#include "ca_midi.h"
646
647struct snd_ca0106;
648
649struct snd_ca0106_channel {
650	struct snd_ca0106 *emu;
651	int number;
652	int use;
653	void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
654	struct snd_ca0106_pcm *epcm;
655};
656
657struct snd_ca0106_pcm {
658	struct snd_ca0106 *emu;
659	struct snd_pcm_substream *substream;
660        int channel_id;
661	unsigned short running;
662};
663
664struct snd_ca0106_details {
665        u32 serial;
666        char * name;
667	int ac97;	/* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
668			   ac97 = 1 -> Default to AC97 in. */
669	int gpio_type;	/* gpio_type = 1 -> shared mic-in/line-in
670			   gpio_type = 2 -> shared side-out/line-in. */
671	int i2c_adc;	/* with i2c_adc=1, the driver adds some capture volume
672			   controls, phone, mic, line-in and aux. */
673	u16 spi_dac;	/* spi_dac = 0 -> no spi interface for DACs
674			   spi_dac = 0x<front><rear><center-lfe><side>
675			   -> specifies DAC id for each channel pair. */
676};
677
678// definition of the chip-specific record
679struct snd_ca0106 {
680	struct snd_card *card;
681	struct snd_ca0106_details *details;
682	struct pci_dev *pci;
683
684	unsigned long port;
685	struct resource *res_port;
686	int irq;
687
688	unsigned int serial;            /* serial number */
689	unsigned short model;		/* subsystem id */
690
691	spinlock_t emu_lock;
692
693	struct snd_ac97 *ac97;
694	struct snd_pcm *pcm[4];
695
696	struct snd_ca0106_channel playback_channels[4];
697	struct snd_ca0106_channel capture_channels[4];
698	u32 spdif_bits[4];             /* s/pdif out default setup */
699	u32 spdif_str_bits[4];         /* s/pdif out per-stream setup */
700	int spdif_enable;
701	int capture_source;
702	int i2c_capture_source;
703	u8 i2c_capture_volume[4][2];
704	int capture_mic_line_in;
705
706	struct snd_dma_buffer buffer;
707
708	struct snd_ca_midi midi;
709	struct snd_ca_midi midi2;
710
711	u16 spi_dac_reg[16];
712
713#ifdef CONFIG_PM_SLEEP
714#define NUM_SAVED_VOLUMES	9
715	unsigned int saved_vol[NUM_SAVED_VOLUMES];
716#endif
717};
718
719int snd_ca0106_mixer(struct snd_ca0106 *emu);
720int snd_ca0106_proc_init(struct snd_ca0106 * emu);
721
722unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, 
723				 unsigned int reg, 
724				 unsigned int chn);
725
726void snd_ca0106_ptr_write(struct snd_ca0106 *emu, 
727			  unsigned int reg, 
728			  unsigned int chn, 
729			  unsigned int data);
730
731int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
732
733int snd_ca0106_spi_write(struct snd_ca0106 * emu,
734				   unsigned int data);
735
736#ifdef CONFIG_PM_SLEEP
737void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
738void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
739#else
740#define snd_ca0106_mixer_suspend(chip)	do { } while (0)
741#define snd_ca0106_mixer_resume(chip)	do { } while (0)
742#endif