diff options
author | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 18:20:36 -0400 |
---|---|---|
committer | Linus Torvalds <torvalds@ppc970.osdl.org> | 2005-04-16 18:20:36 -0400 |
commit | 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch) | |
tree | 0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/pci/ca0106/ca0106.h |
Linux-2.6.12-rc2v2.6.12-rc2
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.
Let it rip!
Diffstat (limited to 'sound/pci/ca0106/ca0106.h')
-rw-r--r-- | sound/pci/ca0106/ca0106.h | 549 |
1 files changed, 549 insertions, 0 deletions
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h new file mode 100644 index 000000000000..deb028851056 --- /dev/null +++ b/sound/pci/ca0106/ca0106.h | |||
@@ -0,0 +1,549 @@ | |||
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: 0.0.20 | ||
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 | * | ||
49 | * | ||
50 | * This code was initally based on code from ALSA's emu10k1x.c which is: | ||
51 | * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> | ||
52 | * | ||
53 | * This program is free software; you can redistribute it and/or modify | ||
54 | * it under the terms of the GNU General Public License as published by | ||
55 | * the Free Software Foundation; either version 2 of the License, or | ||
56 | * (at your option) any later version. | ||
57 | * | ||
58 | * This program is distributed in the hope that it will be useful, | ||
59 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
60 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | ||
61 | * GNU General Public License for more details. | ||
62 | * | ||
63 | * You should have received a copy of the GNU General Public License | ||
64 | * along with this program; if not, write to the Free Software | ||
65 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | ||
66 | * | ||
67 | */ | ||
68 | |||
69 | /************************************************************************************************/ | ||
70 | /* PCI function 0 registers, address = <val> + PCIBASE0 */ | ||
71 | /************************************************************************************************/ | ||
72 | |||
73 | #define PTR 0x00 /* Indexed register set pointer register */ | ||
74 | /* NOTE: The CHANNELNUM and ADDRESS words can */ | ||
75 | /* be modified independently of each other. */ | ||
76 | /* CNL[1:0], ADDR[27:16] */ | ||
77 | |||
78 | #define DATA 0x04 /* Indexed register set data register */ | ||
79 | /* DATA[31:0] */ | ||
80 | |||
81 | #define IPR 0x08 /* Global interrupt pending register */ | ||
82 | /* Clear pending interrupts by writing a 1 to */ | ||
83 | /* the relevant bits and zero to the other bits */ | ||
84 | #define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ | ||
85 | #define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ | ||
86 | #define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ | ||
87 | #define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ | ||
88 | #define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ | ||
89 | #define IPR_SPI 0x00000800 /* SPI transaction completed */ | ||
90 | #define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ | ||
91 | #define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ | ||
92 | #define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */ | ||
93 | #define IPR_GPI 0x00000080 /* General Purpose input changed */ | ||
94 | #define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */ | ||
95 | #define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ | ||
96 | #define IPR_TIMER2 0x00000010 /* 192000Hz Timer */ | ||
97 | #define IPR_TIMER1 0x00000008 /* 44100Hz Timer */ | ||
98 | #define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ | ||
99 | #define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ | ||
100 | #define IPR_PCI 0x00000001 /* PCI Bus error */ | ||
101 | |||
102 | #define INTE 0x0c /* Interrupt enable register */ | ||
103 | |||
104 | #define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ | ||
105 | #define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ | ||
106 | #define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ | ||
107 | #define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ | ||
108 | #define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ | ||
109 | #define INTE_SPI 0x00000800 /* SPI transaction completed */ | ||
110 | #define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ | ||
111 | #define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ | ||
112 | #define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */ | ||
113 | #define INTE_GPI 0x00000080 /* General Purpose input changed */ | ||
114 | #define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */ | ||
115 | #define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ | ||
116 | #define INTE_TIMER2 0x00000010 /* 192000Hz Timer */ | ||
117 | #define INTE_TIMER1 0x00000008 /* 44100Hz Timer */ | ||
118 | #define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ | ||
119 | #define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ | ||
120 | #define INTE_PCI 0x00000001 /* PCI Bus error */ | ||
121 | |||
122 | #define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ | ||
123 | #define HCFG 0x14 /* Hardware config register */ | ||
124 | /* 0x1000 causes AC3 to fails. It adds a dither bit. */ | ||
125 | |||
126 | #define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ | ||
127 | #define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */ | ||
128 | #define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */ | ||
129 | #define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */ | ||
130 | #define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */ | ||
131 | #define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */ | ||
132 | #define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */ | ||
133 | #define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */ | ||
134 | #define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */ | ||
135 | #define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/ | ||
136 | #define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/ | ||
137 | #define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */ | ||
138 | #define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */ | ||
139 | #define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */ | ||
140 | #define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */ | ||
141 | /* NOTE: This should generally never be used. */ | ||
142 | #define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */ | ||
143 | /* NOTE: This should generally never be used. */ | ||
144 | #define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ | ||
145 | /* Should be set to 1 when the EMU10K1 is */ | ||
146 | /* completely initialized. */ | ||
147 | #define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ | ||
148 | /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ | ||
149 | /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ | ||
150 | /* SB Live 24bit: | ||
151 | * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in. | ||
152 | * bit 9 0 = Mute / 1 = Analog out. | ||
153 | * bit 10 0 = Line-in / 1 = Mic-in. | ||
154 | * bit 11 0 = ? / 1 = ? | ||
155 | * bit 12 0 = ? / 1 = ? | ||
156 | * bit 13 0 = ? / 1 = ? | ||
157 | * bit 14 0 = Mute / 1 = Analog out | ||
158 | * bit 15 0 = ? / 1 = ? | ||
159 | * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit. | ||
160 | */ | ||
161 | /* 8 general purpose programmable In/Out pins. | ||
162 | * GPI [8:0] Read only. Default 0. | ||
163 | * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) | ||
164 | * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. | ||
165 | */ | ||
166 | #define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ | ||
167 | |||
168 | #define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ | ||
169 | |||
170 | /********************************************************************************************************/ | ||
171 | /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ | ||
172 | /********************************************************************************************************/ | ||
173 | |||
174 | /* Initally all registers from 0x00 to 0x3f have zero contents. */ | ||
175 | #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ | ||
176 | /* One list entry: 4 bytes for DMA address, | ||
177 | * 4 bytes for period_size << 16. | ||
178 | * One list entry is 8 bytes long. | ||
179 | * One list entry for each period in the buffer. | ||
180 | */ | ||
181 | /* ADDR[31:0], Default: 0x0 */ | ||
182 | #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ | ||
183 | /* SIZE[21:16], Default: 0x8 */ | ||
184 | #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ | ||
185 | /* PTR[5:0], Default: 0x0 */ | ||
186 | #define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ | ||
187 | #define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ | ||
188 | /* DMA[31:0], Default: 0x0 */ | ||
189 | #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ | ||
190 | /* SIZE[31:16], Default: 0x0 */ | ||
191 | #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ | ||
192 | /* POINTER[15:0], Default: 0x0 */ | ||
193 | #define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */ | ||
194 | /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */ | ||
195 | #define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */ | ||
196 | /* Cache size valid [5:0] */ | ||
197 | #define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */ | ||
198 | #define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ | ||
199 | /* DMA[31:0], Default: 0x0 */ | ||
200 | #define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ | ||
201 | /* SIZE[31:16], Default: 0x0 */ | ||
202 | #define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ | ||
203 | /* POINTER[15:0], Default: 0x0 */ | ||
204 | #define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */ | ||
205 | /* Cache size valid [5:0] */ | ||
206 | #define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */ | ||
207 | /* 0x21 - 0x3f unused */ | ||
208 | #define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ | ||
209 | /* Playback (0x1<<channel_id) */ | ||
210 | /* Capture (0x100<<channel_id) */ | ||
211 | /* Playback sample rate 96000 = 0x20000 */ | ||
212 | /* Start Playback [3:0] (one bit per channel) | ||
213 | * Start Capture [11:8] (one bit per channel) | ||
214 | * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) | ||
215 | * Playback mixer in enable [27:24] (one bit per channel) | ||
216 | * Playback mixer out enable [31:28] (one bit per channel) | ||
217 | */ | ||
218 | /* The Digital out jack is shared with the Center/LFE Analogue output. | ||
219 | * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 | ||
220 | * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground | ||
221 | * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. | ||
222 | * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. | ||
223 | * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. | ||
224 | */ | ||
225 | /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS | ||
226 | * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS | ||
227 | * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM. | ||
228 | * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output | ||
229 | */ | ||
230 | /* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. | ||
231 | * 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. | ||
232 | */ | ||
233 | #define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ | ||
234 | #define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ | ||
235 | #define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */ | ||
236 | #define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */ | ||
237 | /* When Channel set to 0: */ | ||
238 | #define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */ | ||
239 | #define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */ | ||
240 | #define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */ | ||
241 | #define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */ | ||
242 | #define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */ | ||
243 | #define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */ | ||
244 | #define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */ | ||
245 | #define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */ | ||
246 | #define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */ | ||
247 | #define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */ | ||
248 | #define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */ | ||
249 | #define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */ | ||
250 | #define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */ | ||
251 | #define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */ | ||
252 | #define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */ | ||
253 | #define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */ | ||
254 | #define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */ | ||
255 | #define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */ | ||
256 | #define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */ | ||
257 | #define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */ | ||
258 | #define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */ | ||
259 | #define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */ | ||
260 | #define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */ | ||
261 | |||
262 | /* When Channel set to 1: */ | ||
263 | #define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */ | ||
264 | #define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */ | ||
265 | #define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */ | ||
266 | #define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */ | ||
267 | #define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */ | ||
268 | #define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ | ||
269 | #define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ | ||
270 | #define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ | ||
271 | #define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ | ||
272 | #define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ | ||
273 | #define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ | ||
274 | #define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ | ||
275 | #define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */ | ||
276 | #define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */ | ||
277 | #define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */ | ||
278 | #define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */ | ||
279 | #define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */ | ||
280 | #define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */ | ||
281 | #define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */ | ||
282 | #define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */ | ||
283 | #define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */ | ||
284 | #define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */ | ||
285 | #define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */ | ||
286 | #define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */ | ||
287 | #define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */ | ||
288 | #define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */ | ||
289 | #define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */ | ||
290 | #define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */ | ||
291 | #define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */ | ||
292 | |||
293 | #define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */ | ||
294 | /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE. | ||
295 | * But as the jack is shared, use 0xf00. | ||
296 | * The Windows2000 driver uses 0x0000000f for both digital and analog. | ||
297 | * 0xf00 introduces interesting noises onto the Center/LFE. | ||
298 | * If you turn the volume up, you hear computer noise, | ||
299 | * e.g. mouse moving, changing between app windows etc. | ||
300 | * So, I am going to set this to 0x0000000f all the time now, | ||
301 | * same as the windows driver does. | ||
302 | * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog. | ||
303 | */ | ||
304 | /* When Channel = 0: | ||
305 | * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit) | ||
306 | * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate) | ||
307 | * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass) | ||
308 | */ | ||
309 | /* When Channel = 1: | ||
310 | * SPDIF 0 User data [7:0] | ||
311 | * SPDIF 1 User data [15:8] | ||
312 | * SPDIF 0 User data [23:16] | ||
313 | * SPDIF 0 User data [31:24] | ||
314 | * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts. | ||
315 | */ | ||
316 | #define WATERMARK 0x46 /* Test bit to indicate cache usage level */ | ||
317 | #define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS. | ||
318 | * When Channel = 0: Bits the same as SPCS channel 0. | ||
319 | * When Channel = 1: Bits the same as SPCS channel 1. | ||
320 | * When Channel = 2: | ||
321 | * SPDIF Input User data [16:0] | ||
322 | * SPDIF Input Frame count [21:16] | ||
323 | */ | ||
324 | #define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */ | ||
325 | #define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */ | ||
326 | #define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */ | ||
327 | #define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */ | ||
328 | #define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ | ||
329 | #define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */ | ||
330 | #define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */ | ||
331 | /* 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 | ||
332 | * Record source select for channel 0 [18:16] | ||
333 | * Record source select for channel 1 [22:20] | ||
334 | * Record source select for channel 2 [26:24] | ||
335 | * Record source select for channel 3 [30:28] | ||
336 | * 0 - SPDIF mixer output. | ||
337 | * 1 - i2s mixer output. | ||
338 | * 2 - SPDIF input. | ||
339 | * 3 - i2s input. | ||
340 | * 4 - AC97 capture. | ||
341 | * 5 - SRC output. | ||
342 | */ | ||
343 | #define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */ | ||
344 | #define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */ | ||
345 | |||
346 | #define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */ | ||
347 | #define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */ | ||
348 | #define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */ | ||
349 | #define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */ | ||
350 | #define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */ | ||
351 | /* Channel_id's handle stereo channels. Channel X is a single mono channel */ | ||
352 | /* Host is input from the PCI bus. */ | ||
353 | /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. | ||
354 | * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. | ||
355 | * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. | ||
356 | * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. | ||
357 | * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. | ||
358 | * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. | ||
359 | * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. | ||
360 | * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. | ||
361 | */ | ||
362 | |||
363 | #define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */ | ||
364 | /* SRC is input from the capture inputs. */ | ||
365 | /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. | ||
366 | * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. | ||
367 | * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. | ||
368 | * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. | ||
369 | * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. | ||
370 | * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. | ||
371 | * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. | ||
372 | * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. | ||
373 | */ | ||
374 | |||
375 | #define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */ | ||
376 | /* SPDIF Mixer input control: | ||
377 | * Invert SRC to SPDIF Mixer [7-0] (One bit per channel) | ||
378 | * Invert Host to SPDIF Mixer [15:8] (One bit per channel) | ||
379 | * SRC to SPDIF Mixer disable [23:16] (One bit per channel) | ||
380 | * Host to SPDIF Mixer disable [31:24] (One bit per channel) | ||
381 | */ | ||
382 | #define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */ | ||
383 | /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */ | ||
384 | /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */ | ||
385 | /* One register for each of the 4 stereo streams. */ | ||
386 | /* SRC Right volume [7:0] | ||
387 | * SRC Left volume [15:8] | ||
388 | * Host Right volume [23:16] | ||
389 | * Host Left volume [31:24] | ||
390 | */ | ||
391 | #define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */ | ||
392 | /* 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. */ | ||
393 | #define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */ | ||
394 | /* 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. */ | ||
395 | #define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */ | ||
396 | /* 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. */ | ||
397 | #define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */ | ||
398 | /* 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. */ | ||
399 | #define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */ | ||
400 | #define UART_A_DATA 0x6c /* Uart, used in setting sample rates, bits per sample etc. */ | ||
401 | #define UART_A_CMD 0x6d /* Uart, used in setting sample rates, bits per sample etc. */ | ||
402 | #define UART_B_DATA 0x6e /* Uart, Unknown. */ | ||
403 | #define UART_B_CMD 0x6f /* Uart, Unknown. */ | ||
404 | #define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */ | ||
405 | /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0 | ||
406 | * Rate Locked [20] | ||
407 | * SPDIF Locked [21] For SPDIF channel only. | ||
408 | * Valid Audio [22] For SPDIF channel only. | ||
409 | */ | ||
410 | #define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */ | ||
411 | /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */ | ||
412 | /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */ | ||
413 | /* Sample rate output control register Channel=0 | ||
414 | * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) | ||
415 | * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) | ||
416 | * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. | ||
417 | * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) | ||
418 | * Record mixer output enable [12:10] | ||
419 | * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) | ||
420 | * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) | ||
421 | * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) | ||
422 | * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0) | ||
423 | * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.) | ||
424 | * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.) | ||
425 | * I2S input mode [23] (0=Slave, 1=Master) | ||
426 | * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) | ||
427 | * SPDIF output source select [26] (0=host, 1=SRC) | ||
428 | * Not used [27] | ||
429 | * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) | ||
430 | * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) | ||
431 | */ | ||
432 | /* Sample rate output control register Channel=1 | ||
433 | * I2S Input 0 volume Right [7:0] | ||
434 | * I2S Input 0 volume Left [15:8] | ||
435 | * I2S Input 1 volume Right [23:16] | ||
436 | * I2S Input 1 volume Left [31:24] | ||
437 | */ | ||
438 | /* Sample rate output control register Channel=2 | ||
439 | * SPDIF Input volume Right [23:16] | ||
440 | * SPDIF Input volume Left [31:24] | ||
441 | */ | ||
442 | /* Sample rate output control register Channel=3 | ||
443 | * No used | ||
444 | */ | ||
445 | #define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */ | ||
446 | #define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */ | ||
447 | #define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */ | ||
448 | #define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */ | ||
449 | /* Audio output control | ||
450 | * AC97 output enable [5:0] | ||
451 | * I2S output enable [19:16] | ||
452 | * SPDIF output enable [27:24] | ||
453 | */ | ||
454 | #define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ | ||
455 | #define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ | ||
456 | #define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ | ||
457 | /* Sets which Interrupts are enabled. */ | ||
458 | /* 0x00000001 = Half period. Playback. | ||
459 | * 0x00000010 = Full period. Playback. | ||
460 | * 0x00000100 = Half buffer. Playback. | ||
461 | * 0x00001000 = Full buffer. Playback. | ||
462 | * 0x00010000 = Half buffer. Capture. | ||
463 | * 0x00100000 = Full buffer. Capture. | ||
464 | * Capture can only do 2 periods. | ||
465 | * 0x01000000 = End audio. Playback. | ||
466 | * 0x40000000 = Half buffer Playback,Caputre xrun. | ||
467 | * 0x80000000 = Full buffer Playback,Caputre xrun. | ||
468 | */ | ||
469 | #define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */ | ||
470 | /* Shows which interrupts are active at the moment. */ | ||
471 | /* Same bit layout as EXTENDED_INT_MASK */ | ||
472 | #define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */ | ||
473 | #define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */ | ||
474 | #define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */ | ||
475 | /* Causes interrupts based on timer intervals. */ | ||
476 | #define SPI 0x7a /* SPI: Serial Interface Register */ | ||
477 | #define I2C_A 0x7b /* I2C Address. 32 bit */ | ||
478 | #define I2C_0 0x7c /* I2C Data Port 0. 32 bit */ | ||
479 | #define I2C_1 0x7d /* I2C Data Port 1. 32 bit */ | ||
480 | |||
481 | |||
482 | #define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ | ||
483 | #define PCM_FRONT_CHANNEL 0 | ||
484 | #define PCM_REAR_CHANNEL 1 | ||
485 | #define PCM_CENTER_LFE_CHANNEL 2 | ||
486 | #define PCM_UNKNOWN_CHANNEL 3 | ||
487 | #define CONTROL_FRONT_CHANNEL 0 | ||
488 | #define CONTROL_REAR_CHANNEL 3 | ||
489 | #define CONTROL_CENTER_LFE_CHANNEL 1 | ||
490 | #define CONTROL_UNKNOWN_CHANNEL 2 | ||
491 | |||
492 | typedef struct snd_ca0106_channel ca0106_channel_t; | ||
493 | typedef struct snd_ca0106 ca0106_t; | ||
494 | typedef struct snd_ca0106_pcm ca0106_pcm_t; | ||
495 | |||
496 | struct snd_ca0106_channel { | ||
497 | ca0106_t *emu; | ||
498 | int number; | ||
499 | int use; | ||
500 | void (*interrupt)(ca0106_t *emu, ca0106_channel_t *channel); | ||
501 | ca0106_pcm_t *epcm; | ||
502 | }; | ||
503 | |||
504 | struct snd_ca0106_pcm { | ||
505 | ca0106_t *emu; | ||
506 | snd_pcm_substream_t *substream; | ||
507 | int channel_id; | ||
508 | unsigned short running; | ||
509 | }; | ||
510 | |||
511 | // definition of the chip-specific record | ||
512 | struct snd_ca0106 { | ||
513 | snd_card_t *card; | ||
514 | struct pci_dev *pci; | ||
515 | |||
516 | unsigned long port; | ||
517 | struct resource *res_port; | ||
518 | int irq; | ||
519 | |||
520 | unsigned int revision; /* chip revision */ | ||
521 | unsigned int serial; /* serial number */ | ||
522 | unsigned short model; /* subsystem id */ | ||
523 | |||
524 | spinlock_t emu_lock; | ||
525 | |||
526 | ac97_t *ac97; | ||
527 | snd_pcm_t *pcm; | ||
528 | |||
529 | ca0106_channel_t playback_channels[4]; | ||
530 | ca0106_channel_t capture_channels[4]; | ||
531 | u32 spdif_bits[4]; /* s/pdif out setup */ | ||
532 | int spdif_enable; | ||
533 | int capture_source; | ||
534 | |||
535 | struct snd_dma_buffer buffer; | ||
536 | }; | ||
537 | |||
538 | int __devinit snd_ca0106_mixer(ca0106_t *emu); | ||
539 | int __devinit snd_ca0106_proc_init(ca0106_t * emu); | ||
540 | |||
541 | unsigned int snd_ca0106_ptr_read(ca0106_t * emu, | ||
542 | unsigned int reg, | ||
543 | unsigned int chn); | ||
544 | |||
545 | void snd_ca0106_ptr_write(ca0106_t *emu, | ||
546 | unsigned int reg, | ||
547 | unsigned int chn, | ||
548 | unsigned int data); | ||
549 | |||