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
6 * FEATURES currently supported:
7 * See ca0106_main.c for features.
10 * Support interrupts per period.
11 * Removed noise from Center/LFE channel when in Analog mode.
12 * Rename and remove mixer controls.
14 * Use separate card based DMA buffer for periods table list.
16 * Change remove and rename ctrls into lists.
18 * Try to fix capture sources.
21 * Enable S32_LE format support.
23 * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
25 * Add Model name recognition.
27 * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
28 * Remove redundent "voice" handling.
30 * Single trigger call for multi channels.
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.
37 * Separated ca0106.c into separate functional .c files.
39 * Implement 192000 sample rate.
41 * Add support for SB0410 and SB0413.
43 * Modified Copyright message.
45 * Added I2C and SPI registers. Filled in interrupt enable.
47 * Added GPIO info for SB Live 24bit.
49 * Implement support for Line-in capture on SB Live 24bit.
51 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
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>
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.
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.
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
73 /************************************************************************************************/
74 /* PCI function 0 registers, address = <val> + PCIBASE0 */
75 /************************************************************************************************/
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] */
82 #define DATA 0x04 /* Indexed register set data register */
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 */
106 #define INTE 0x0c /* Interrupt enable register */
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 */
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. */
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. */
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.
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.
170 #define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
172 #define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
174 /********************************************************************************************************/
175 /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
176 /********************************************************************************************************/
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.
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)
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.
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
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.
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) */
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 */
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.
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)
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.
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.
324 * SPDIF Input User data [16:0]
325 * SPDIF Input Frame count [21:16]
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.
346 #define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */
347 #define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */
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.
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.
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)
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]
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) */
408 /* unique channel identifier for midi->channel */
410 #define CA0106_MIDI_CHAN_A 0x1
411 #define CA0106_MIDI_CHAN_B 0x2
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
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
424 * SPDIF Locked [21] For SPDIF channel only.
425 * Valid Audio [22] For SPDIF channel only.
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)
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)
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]
455 /* Sample rate output control register Channel=2
456 * SPDIF Input volume Right [23:16]
457 * SPDIF Input volume Left [31:24]
459 /* Sample rate output control register Channel=3
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]
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.
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 */
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
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
512 #define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register
513 #define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register
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
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
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
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
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 */
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
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) */
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 */
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 */
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)
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)
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)
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 */
642 #define SPI_PWRDNALL_REG 10 /* power down everything */
643 #define SPI_PWRDNALL_BIT (1<<4)
649 struct snd_ca0106_channel
{
650 struct snd_ca0106
*emu
;
653 void (*interrupt
)(struct snd_ca0106
*emu
, struct snd_ca0106_channel
*channel
);
654 struct snd_ca0106_pcm
*epcm
;
657 struct snd_ca0106_pcm
{
658 struct snd_ca0106
*emu
;
659 struct snd_pcm_substream
*substream
;
661 unsigned short running
;
664 struct snd_ca0106_details
{
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. */
678 // definition of the chip-specific record
680 struct snd_card
*card
;
681 struct snd_ca0106_details
*details
;
685 struct resource
*res_port
;
688 unsigned int serial
; /* serial number */
689 unsigned short model
; /* subsystem id */
693 struct snd_ac97
*ac97
;
694 struct snd_pcm
*pcm
[4];
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 */
702 int i2c_capture_source
;
703 u8 i2c_capture_volume
[4][2];
704 int capture_mic_line_in
;
706 struct snd_dma_buffer buffer
;
708 struct snd_ca_midi midi
;
709 struct snd_ca_midi midi2
;
714 #define NUM_SAVED_VOLUMES 9
715 unsigned int saved_vol
[NUM_SAVED_VOLUMES
];
719 int snd_ca0106_mixer(struct snd_ca0106
*emu
);
720 int snd_ca0106_proc_init(struct snd_ca0106
* emu
);
722 unsigned int snd_ca0106_ptr_read(struct snd_ca0106
* emu
,
726 void snd_ca0106_ptr_write(struct snd_ca0106
*emu
,
731 int snd_ca0106_i2c_write(struct snd_ca0106
*emu
, u32 reg
, u32 value
);
733 int snd_ca0106_spi_write(struct snd_ca0106
* emu
,
737 void snd_ca0106_mixer_suspend(struct snd_ca0106
*chip
);
738 void snd_ca0106_mixer_resume(struct snd_ca0106
*chip
);
740 #define snd_ca0106_mixer_suspend(chip) do { } while (0)
741 #define snd_ca0106_mixer_resume(chip) do { } while (0)