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 CA0106_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 CA0106_DATA 0x04 /* Indexed register set data register */ |
68 | /* DATA[31:0] */ |
69 | |
70 | #define CA0106_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 CA0106_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 CA0106_UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ |
112 | #define CA0106_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 CA0106_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 CA0106_AC97DATA 0x1c /* AC97 register set data register (16 bit) */ |
156 | |
157 | #define CA0106_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 | |
632 | struct snd_ca0106; |
633 | |
634 | struct 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 | |
642 | struct snd_ca0106_pcm { |
643 | struct snd_ca0106 *emu; |
644 | struct snd_pcm_substream *substream; |
645 | int channel_id; |
646 | unsigned short running; |
647 | }; |
648 | |
649 | struct 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 |
664 | struct 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 | int irq; |
671 | |
672 | unsigned int serial; /* serial number */ |
673 | unsigned short model; /* subsystem id */ |
674 | |
675 | spinlock_t emu_lock; |
676 | |
677 | struct snd_ac97 *ac97; |
678 | struct snd_pcm *pcm[4]; |
679 | |
680 | struct snd_ca0106_channel playback_channels[4]; |
681 | struct snd_ca0106_channel capture_channels[4]; |
682 | u32 spdif_bits[4]; /* s/pdif out default setup */ |
683 | u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */ |
684 | int spdif_enable; |
685 | int capture_source; |
686 | int i2c_capture_source; |
687 | u8 i2c_capture_volume[4][2]; |
688 | int capture_mic_line_in; |
689 | |
690 | struct snd_dma_buffer *buffer; |
691 | |
692 | struct snd_ca_midi midi; |
693 | struct snd_ca_midi midi2; |
694 | |
695 | u16 spi_dac_reg[16]; |
696 | |
697 | #ifdef CONFIG_PM_SLEEP |
698 | #define NUM_SAVED_VOLUMES 9 |
699 | unsigned int saved_vol[NUM_SAVED_VOLUMES]; |
700 | #endif |
701 | }; |
702 | |
703 | int snd_ca0106_mixer(struct snd_ca0106 *emu); |
704 | int snd_ca0106_proc_init(struct snd_ca0106 * emu); |
705 | |
706 | unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, |
707 | unsigned int reg, |
708 | unsigned int chn); |
709 | |
710 | void snd_ca0106_ptr_write(struct snd_ca0106 *emu, |
711 | unsigned int reg, |
712 | unsigned int chn, |
713 | unsigned int data); |
714 | |
715 | int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); |
716 | |
717 | int snd_ca0106_spi_write(struct snd_ca0106 * emu, |
718 | unsigned int data); |
719 | |
720 | #ifdef CONFIG_PM_SLEEP |
721 | void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); |
722 | void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); |
723 | #else |
724 | #define snd_ca0106_mixer_suspend(chip) do { } while (0) |
725 | #define snd_ca0106_mixer_resume(chip) do { } while (0) |
726 | #endif |
727 | |