Adding upstream version 6.6.15.upstream/6.6.15

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

7 files changed, 4269 insertions, 0 deletions

diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile
new file mode 100644
index 0000000000..9e51d3df3e
--- /dev/null
+++ b/sound/pci/ca0106/Makefile
@@ -0,0 +1,5 @@
+# SPDX-License-Identifier: GPL-2.0-only
+snd-ca0106-objs := ca0106_main.o ca0106_mixer.o ca_midi.o
+snd-ca0106-$(CONFIG_SND_PROC_FS) += ca0106_proc.o
+
+obj-$(CONFIG_SND_CA0106) += snd-ca0106.o
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
new file mode 100644
index 0000000000..991b1c5d41
--- /dev/null
+++ b/sound/pci/ca0106/ca0106.h
@@ -0,0 +1,726 @@
+/* SPDX-License-Identifier: GPL-2.0-or-later */
+/*
+ *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ *  Version: 0.0.22
+ *
+ *  FEATURES currently supported:
+ *    See ca0106_main.c for features.
+ * 
+ *  Changelog:
+ *    Support interrupts per period.
+ *    Removed noise from Center/LFE channel when in Analog mode.
+ *    Rename and remove mixer controls.
+ *  0.0.6
+ *    Use separate card based DMA buffer for periods table list.
+ *  0.0.7
+ *    Change remove and rename ctrls into lists.
+ *  0.0.8
+ *    Try to fix capture sources.
+ *  0.0.9
+ *    Fix AC3 output.
+ *    Enable S32_LE format support.
+ *  0.0.10
+ *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ *  0.0.11
+ *    Add Model name recognition.
+ *  0.0.12
+ *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ *    Remove redundent "voice" handling.
+ *  0.0.13
+ *    Single trigger call for multi channels.
+ *  0.0.14
+ *    Set limits based on what the sound card hardware can do.
+ *    playback periods_min=2, periods_max=8
+ *    capture hw constraints require period_size = n * 64 bytes.
+ *    playback hw constraints require period_size = n * 64 bytes.
+ *  0.0.15
+ *    Separated ca0106.c into separate functional .c files.
+ *  0.0.16
+ *    Implement 192000 sample rate.
+ *  0.0.17
+ *    Add support for SB0410 and SB0413.
+ *  0.0.18
+ *    Modified Copyright message.
+ *  0.0.19
+ *    Added I2C and SPI registers. Filled in interrupt enable.
+ *  0.0.20
+ *    Added GPIO info for SB Live 24bit.
+ *  0.0.21
+ *   Implement support for Line-in capture on SB Live 24bit.
+ *  0.0.22
+ *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
+ *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ */
+
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0						*/
+/************************************************************************************************/
+
+#define CA0106_PTR		0x00		/* Indexed register set pointer register	*/
+						/* NOTE: The CHANNELNUM and ADDRESS words can	*/
+						/* be modified independently of each other.	*/
+						/* CNL[1:0], ADDR[27:16]                        */
+
+#define CA0106_DATA		0x04		/* Indexed register set data register		*/
+						/* DATA[31:0]					*/
+
+#define CA0106_IPR		0x08		/* Global interrupt pending register		*/
+						/* Clear pending interrupts by writing a 1 to	*/
+						/* the relevant bits and zero to the other bits	*/
+#define IPR_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
+#define IPR_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
+#define IPR_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
+#define IPR_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
+#define IPR_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
+#define IPR_SPI			0x00000800      /* SPI transaction completed			*/
+#define IPR_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
+#define IPR_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
+#define IPR_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x76	*/
+#define IPR_GPI			0x00000080      /* General Purpose input changed		*/
+#define IPR_SRC_LOCKED          0x00000040      /* SRC lock status changed			*/
+#define IPR_SPDIF_STATUS        0x00000020      /* SPDIF status changed				*/
+#define IPR_TIMER2              0x00000010      /* 192000Hz Timer				*/
+#define IPR_TIMER1              0x00000008      /* 44100Hz Timer				*/
+#define IPR_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
+#define IPR_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
+#define IPR_PCI			0x00000001	/* PCI Bus error				*/
+
+#define CA0106_INTE		0x0c		/* Interrupt enable register			*/
+
+#define INTE_MIDI_RX_B		0x00020000	/* MIDI UART-B Receive buffer non-empty		*/
+#define INTE_MIDI_TX_B		0x00010000	/* MIDI UART-B Transmit buffer empty		*/
+#define INTE_SPDIF_IN_USER	0x00004000      /* SPDIF input user data has 16 more bits	*/
+#define INTE_SPDIF_OUT_USER	0x00002000      /* SPDIF output user data needs 16 more bits	*/
+#define INTE_SPDIF_OUT_FRAME	0x00001000      /* SPDIF frame about to start			*/
+#define INTE_SPI		0x00000800      /* SPI transaction completed			*/
+#define INTE_I2C_EEPROM		0x00000400      /* I2C EEPROM transaction completed		*/
+#define INTE_I2C_DAC		0x00000200      /* I2C DAC transaction completed		*/
+#define INTE_AI			0x00000100      /* Audio pending register changed. See PTR reg 0x75 */
+#define INTE_GPI		0x00000080      /* General Purpose input changed		*/
+#define INTE_SRC_LOCKED         0x00000040      /* SRC lock status changed			*/
+#define INTE_SPDIF_STATUS       0x00000020      /* SPDIF status changed				*/
+#define INTE_TIMER2             0x00000010      /* 192000Hz Timer				*/
+#define INTE_TIMER1             0x00000008      /* 44100Hz Timer				*/
+#define INTE_MIDI_RX_A		0x00000004	/* MIDI UART-A Receive buffer non-empty		*/
+#define INTE_MIDI_TX_A		0x00000002	/* MIDI UART-A Transmit buffer empty		*/
+#define INTE_PCI		0x00000001	/* PCI Bus error				*/
+
+#define CA0106_UNKNOWN10	0x10		/* Unknown ??. Defaults to 0 */
+#define CA0106_HCFG		0x14		/* Hardware config register			*/
+						/* 0x1000 causes AC3 to fails. It adds a dither bit. */
+
+#define HCFG_STAC		0x10000000	/* Special mode for STAC9460 Codec. */
+#define HCFG_CAPTURE_I2S_BYPASS	0x08000000	/* 1 = bypass I2S input async SRC. */
+#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000	/* 1 = bypass SPDIF input async SRC. */
+#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000	/* 0 = I2S IN mixer output, 1 = I2S IN1. */
+#define HCFG_FORCE_LOCK		0x01000000	/* For test only. Force input SRC tracker to lock. */
+#define HCFG_PLAYBACK_ATTENUATION 0x00006000	/* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
+#define HCFG_PLAYBACK_DITHER	0x00001000	/* 1 = Add dither bit to all playback channels. */
+#define HCFG_PLAYBACK_S32_LE	0x00000800	/* 1 = S32_LE, 0 = S16_LE                       */
+#define HCFG_CAPTURE_S32_LE	0x00000400	/* 1 = S32_LE, 0 = S16_LE (S32_LE current not working)	*/
+#define HCFG_8_CHANNEL_PLAY	0x00000200	/* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_8_CHANNEL_CAPTURE	0x00000100	/* 1 = 8 channels, 0 = 2 channels per substream.*/
+#define HCFG_MONO		0x00000080	/* 1 = I2S Input mono                           */
+#define HCFG_I2S_OUTPUT		0x00000010	/* 1 = I2S Output disabled                      */
+#define HCFG_AC97		0x00000008	/* 0 = AC97 1.0, 1 = AC97 2.0                   */
+#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004	/* 1 = Cancel bustmaster accesses to soundcache */
+						/* NOTE: This should generally never be used.  	*/
+#define HCFG_LOCK_CAPTURE_CACHE	0x00000002	/* 1 = Cancel bustmaster accesses to soundcache */
+						/* NOTE: This should generally never be used.  	*/
+#define HCFG_AUDIOENABLE	0x00000001	/* 0 = CODECs transmit zero-valued samples	*/
+						/* Should be set to 1 when the EMU10K1 is	*/
+						/* completely initialized.			*/
+#define CA0106_GPIO		0x18		/* Defaults: 005f03a3-Analog, 005f02a2-SPDIF.   */
+						/* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
+						/* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
+						/* SB Live 24bit:
+						 * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
+						 * bit 9 0 = Mute / 1 = Analog out.
+						 * bit 10 0 = Line-in / 1 = Mic-in.
+						 * bit 11 0 = ? / 1 = ?
+						 * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
+						 * bit 13 0 = ? / 1 = ?
+						 * bit 14 0 = Mute / 1 = Analog out
+						 * bit 15 0 = ? / 1 = ?
+						 * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
+						 */
+						/* 8 general purpose programmable In/Out pins.
+						 * GPI [8:0] Read only. Default 0.
+						 * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
+						 * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
+						 */
+#define CA0106_AC97DATA		0x1c		/* AC97 register set data register (16 bit)	*/
+
+#define CA0106_AC97ADDRESS	0x1e		/* AC97 register set address register (8 bit)	*/
+
+/********************************************************************************************************/
+/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers                     */
+/********************************************************************************************************/
+                                                                                                                           
+/* Initially all registers from 0x00 to 0x3f have zero contents. */
+#define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
+						/* One list entry: 4 bytes for DMA address, 
+						 * 4 bytes for period_size << 16.
+						 * One list entry is 8 bytes long.
+						 * One list entry for each period in the buffer.
+						 */
+						/* ADDR[31:0], Default: 0x0 */
+#define PLAYBACK_LIST_SIZE	0x01		/* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000  */
+						/* SIZE[21:16], Default: 0x8 */
+#define PLAYBACK_LIST_PTR	0x02		/* Pointer to the current period being played */
+						/* PTR[5:0], Default: 0x0 */
+#define PLAYBACK_UNKNOWN3	0x03		/* Not used ?? */
+#define PLAYBACK_DMA_ADDR	0x04		/* Playback DMA address */
+						/* DMA[31:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_SIZE	0x05		/* Playback period size. win2000 uses 0x04000000 */
+						/* SIZE[31:16], Default: 0x0 */
+#define PLAYBACK_POINTER	0x06		/* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
+						/* POINTER[15:0], Default: 0x0 */
+#define PLAYBACK_PERIOD_END_ADDR 0x07		/* Playback fifo end address */
+						/* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
+#define PLAYBACK_FIFO_OFFSET_ADDRESS	0x08	/* Current fifo offset address [21:16] */
+						/* Cache size valid [5:0] */
+#define PLAYBACK_UNKNOWN9	0x09		/* 0x9 to 0xf Unused */
+#define CAPTURE_DMA_ADDR	0x10		/* Capture DMA address */
+						/* DMA[31:0], Default: 0x0 */
+#define CAPTURE_BUFFER_SIZE	0x11		/* Capture buffer size */
+						/* SIZE[31:16], Default: 0x0 */
+#define CAPTURE_POINTER		0x12		/* Capture buffer pointer. Sample currently in ADC */
+						/* POINTER[15:0], Default: 0x0 */
+#define CAPTURE_FIFO_OFFSET_ADDRESS	0x13	/* Current fifo offset address [21:16] */
+						/* Cache size valid [5:0] */
+#define PLAYBACK_LAST_SAMPLE    0x20		/* The sample currently being played */
+/* 0x21 - 0x3f unused */
+#define BASIC_INTERRUPT         0x40		/* Used by both playback and capture interrupt handler */
+						/* Playback (0x1<<channel_id) */
+						/* Capture  (0x100<<channel_id) */
+						/* Playback sample rate 96000 = 0x20000 */
+						/* Start Playback [3:0] (one bit per channel)
+						 * Start Capture [11:8] (one bit per channel)
+						 * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+						 * Playback mixer in enable [27:24] (one bit per channel)
+						 * Playback mixer out enable [31:28] (one bit per channel)
+						 */
+/* The Digital out jack is shared with the Center/LFE Analogue output. 
+ * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
+ * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
+ * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
+ * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
+ * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
+ */
+/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
+ * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
+ * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
+ * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
+ */
+/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
+ * 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. 
+ */
+#define SPCS0			0x41		/* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006	*/
+#define SPCS1			0x42		/* SPDIF output Channel Status 1 register. For Front */
+#define SPCS2			0x43		/* SPDIF output Channel Status 2 register. For Center/LFE */
+#define SPCS3			0x44		/* SPDIF output Channel Status 3 register. Unknown */
+						/* When Channel set to 0: */
+#define SPCS_CLKACCYMASK	0x30000000	/* Clock accuracy				*/
+#define SPCS_CLKACCY_1000PPM	0x00000000	/* 1000 parts per million			*/
+#define SPCS_CLKACCY_50PPM	0x10000000	/* 50 parts per million				*/
+#define SPCS_CLKACCY_VARIABLE	0x20000000	/* Variable accuracy				*/
+#define SPCS_SAMPLERATEMASK	0x0f000000	/* Sample rate					*/
+#define SPCS_SAMPLERATE_44	0x00000000	/* 44.1kHz sample rate				*/
+#define SPCS_SAMPLERATE_48	0x02000000	/* 48kHz sample rate				*/
+#define SPCS_SAMPLERATE_32	0x03000000	/* 32kHz sample rate				*/
+#define SPCS_CHANNELNUMMASK	0x00f00000	/* Channel number				*/
+#define SPCS_CHANNELNUM_UNSPEC	0x00000000	/* Unspecified channel number			*/
+#define SPCS_CHANNELNUM_LEFT	0x00100000	/* Left channel					*/
+#define SPCS_CHANNELNUM_RIGHT	0x00200000	/* Right channel				*/
+#define SPCS_SOURCENUMMASK	0x000f0000	/* Source number				*/
+#define SPCS_SOURCENUM_UNSPEC	0x00000000	/* Unspecified source number			*/
+#define SPCS_GENERATIONSTATUS	0x00008000	/* Originality flag (see IEC-958 spec)		*/
+#define SPCS_CATEGORYCODEMASK	0x00007f00	/* Category code (see IEC-958 spec)		*/
+#define SPCS_MODEMASK		0x000000c0	/* Mode (see IEC-958 spec)			*/
+#define SPCS_EMPHASISMASK	0x00000038	/* Emphasis					*/
+#define SPCS_EMPHASIS_NONE	0x00000000	/* No emphasis					*/
+#define SPCS_EMPHASIS_50_15	0x00000008	/* 50/15 usec 2 channel				*/
+#define SPCS_COPYRIGHT		0x00000004	/* Copyright asserted flag -- do not modify	*/
+#define SPCS_NOTAUDIODATA	0x00000002	/* 0 = Digital audio, 1 = not audio		*/
+#define SPCS_PROFESSIONAL	0x00000001	/* 0 = Consumer (IEC-958), 1 = pro (AES3-1992)	*/
+
+						/* When Channel set to 1: */
+#define SPCS_WORD_LENGTH_MASK	0x0000000f	/* Word Length Mask				*/
+#define SPCS_WORD_LENGTH_16	0x00000008	/* Word Length 16 bit				*/
+#define SPCS_WORD_LENGTH_17	0x00000006	/* Word Length 17 bit				*/
+#define SPCS_WORD_LENGTH_18	0x00000004	/* Word Length 18 bit				*/
+#define SPCS_WORD_LENGTH_19	0x00000002	/* Word Length 19 bit				*/
+#define SPCS_WORD_LENGTH_20A	0x0000000a	/* Word Length 20 bit				*/
+#define SPCS_WORD_LENGTH_20	0x00000009	/* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
+#define SPCS_WORD_LENGTH_21	0x00000007	/* Word Length 21 bit				*/
+#define SPCS_WORD_LENGTH_22	0x00000005	/* Word Length 22 bit				*/
+#define SPCS_WORD_LENGTH_23	0x00000003	/* Word Length 23 bit				*/
+#define SPCS_WORD_LENGTH_24	0x0000000b	/* Word Length 24 bit				*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_MASK	0x000000f0 /* Original Sample rate			*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_NONE	0x00000000 /* Original Sample rate not indicated	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_16000	0x00000010 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES1	0x00000020 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_32000	0x00000030 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_12000	0x00000040 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_11025	0x00000050 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_8000	0x00000060 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_RES2	0x00000070 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_24000	0x00000090 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_96000	0x000000a0 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_48000	0x000000b0 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_22050	0x000000d0 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_88200	0x000000e0 /* Original Sample rate	*/
+#define SPCS_ORIGINAL_SAMPLE_RATE_44100	0x000000f0 /* Original Sample rate	*/
+
+#define SPDIF_SELECT1		0x45		/* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
+						/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
+						 * But as the jack is shared, use 0xf00.
+						 * The Windows2000 driver uses 0x0000000f for both digital and analog.
+						 * 0xf00 introduces interesting noises onto the Center/LFE.
+						 * If you turn the volume up, you hear computer noise,
+						 * e.g. mouse moving, changing between app windows etc.
+						 * So, I am going to set this to 0x0000000f all the time now,
+						 * same as the windows driver does.
+						 * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
+						 */
+						/* When Channel = 0:
+						 * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
+						 * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
+						 * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
+						 */
+						/* When Channel = 1:
+						 * SPDIF 0 User data [7:0]
+						 * SPDIF 1 User data [15:8]
+						 * SPDIF 0 User data [23:16]
+						 * SPDIF 0 User data [31:24]
+						 * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
+						 */
+#define WATERMARK		0x46		/* Test bit to indicate cache usage level */
+#define SPDIF_INPUT_STATUS	0x49		/* SPDIF Input status register. Bits the same as SPCS.
+						 * When Channel = 0: Bits the same as SPCS channel 0.
+						 * When Channel = 1: Bits the same as SPCS channel 1.
+						 * When Channel = 2:
+						 * SPDIF Input User data [16:0]
+						 * SPDIF Input Frame count [21:16]
+						 */
+#define CAPTURE_CACHE_DATA	0x50		/* 0x50-0x5f Recorded samples. */
+#define CAPTURE_SOURCE          0x60            /* Capture Source 0 = MIC */
+#define CAPTURE_SOURCE_CHANNEL0 0xf0000000	/* Mask for selecting the Capture sources */
+#define CAPTURE_SOURCE_CHANNEL1 0x0f000000	/* 0 - SPDIF mixer output. */
+#define CAPTURE_SOURCE_CHANNEL2 0x00f00000      /* 1 - What you hear or . 2 - ?? */
+#define CAPTURE_SOURCE_CHANNEL3 0x000f0000	/* 3 - Mic in, Line in, TAD in, Aux in. */
+#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff	/* Default 0x00e4 */
+						/* 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 
+						 * Record source select for channel 0 [18:16]
+						 * Record source select for channel 1 [22:20]
+						 * Record source select for channel 2 [26:24]
+						 * Record source select for channel 3 [30:28]
+						 * 0 - SPDIF mixer output.
+						 * 1 - i2s mixer output.
+						 * 2 - SPDIF input.
+						 * 3 - i2s input.
+						 * 4 - AC97 capture.
+						 * 5 - SRC output.
+						 */
+#define CAPTURE_VOLUME1         0x61            /* Capture  volume per channel 0-3 */
+#define CAPTURE_VOLUME2         0x62            /* Capture  volume per channel 4-7 */
+
+#define PLAYBACK_ROUTING1       0x63            /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
+#define ROUTING1_REAR           0x77000000      /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
+#define ROUTING1_NULL           0x00770000      /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
+#define ROUTING1_CENTER_LFE     0x00007700      /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
+#define ROUTING1_FRONT          0x00000077	/* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
+						/* Channel_id's handle stereo channels. Channel X is a single mono channel */
+						/* Host is input from the PCI bus. */
+						/* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+						 * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+						 */
+
+#define PLAYBACK_ROUTING2       0x64            /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
+						/* SRC is input from the capture inputs. */
+						/* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
+						 * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
+						 */
+
+#define PLAYBACK_MUTE           0x65            /* Unknown. While playing 0x0, while silent 0x00fc0000 */
+						/* SPDIF Mixer input control:
+						 * Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
+						 * Invert Host to SPDIF Mixer [15:8] (One bit per channel)
+						 * SRC to SPDIF Mixer disable [23:16] (One bit per channel)
+						 * Host to SPDIF Mixer disable [31:24] (One bit per channel)
+						 */
+#define PLAYBACK_VOLUME1        0x66            /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
+						/* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
+						/* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
+						/* One register for each of the 4 stereo streams. */
+						/* SRC Right volume [7:0]
+						 * SRC Left  volume [15:8]
+						 * Host Right volume [23:16]
+						 * Host Left  volume [31:24]
+						 */
+#define CAPTURE_ROUTING1        0x67            /* Capture Routing. Default 0x32765410 */
+						/* 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. */
+#define CAPTURE_ROUTING2        0x68            /* Unknown Routing. Default 0x76767676 */
+						/* 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. */
+#define CAPTURE_MUTE            0x69            /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
+						/* 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. */
+#define PLAYBACK_VOLUME2        0x6a            /* Playback Analog volume per channel. Does not effect AC3 output */
+						/* 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. */
+#define UNKNOWN6b               0x6b            /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
+#define MIDI_UART_A_DATA		0x6c            /* Midi Uart A Data */
+#define MIDI_UART_A_CMD		0x6d            /* Midi Uart A Command/Status */
+#define MIDI_UART_B_DATA		0x6e            /* Midi Uart B Data (currently unused) */
+#define MIDI_UART_B_CMD		0x6f            /* Midi Uart B Command/Status (currently unused) */
+
+/* unique channel identifier for midi->channel */
+
+#define CA0106_MIDI_CHAN_A		0x1
+#define CA0106_MIDI_CHAN_B		0x2
+
+/* from mpu401 */
+
+#define CA0106_MIDI_INPUT_AVAIL 	0x80
+#define CA0106_MIDI_OUTPUT_READY	0x40
+#define CA0106_MPU401_RESET		0xff
+#define CA0106_MPU401_ENTER_UART	0x3f
+#define CA0106_MPU401_ACK		0xfe
+
+#define SAMPLE_RATE_TRACKER_STATUS 0x70         /* Readonly. Default 00108000 00108000 00500000 00500000 */
+						/* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 =  1.0
+						 * Rate Locked [20]
+						 * SPDIF Locked [21] For SPDIF channel only.
+						 * Valid Audio [22] For SPDIF channel only.
+						 */
+#define CAPTURE_CONTROL         0x71            /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
+						/* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
+						/* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
+						/* Sample rate output control register Channel=0
+						 * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+						 * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+						 * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
+						 * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
+						 * Record mixer output enable [12:10] 
+						 * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+						 * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+						 * I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
+						 * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
+						 * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
+						 * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
+						 * I2S input mode [23] (0=Slave, 1=Master)
+						 * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
+						 * SPDIF output source select [26] (0=host, 1=SRC)
+						 * Not used [27]
+						 * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+						 * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
+						 */ 
+						/* Sample rate output control register Channel=1
+						 * I2S Input 0 volume Right [7:0]
+						 * I2S Input 0 volume Left [15:8]
+						 * I2S Input 1 volume Right [23:16]
+						 * I2S Input 1 volume Left [31:24]
+						 */
+						/* Sample rate output control register Channel=2
+						 * SPDIF Input volume Right [23:16]
+						 * SPDIF Input volume Left [31:24]
+						 */
+						/* Sample rate output control register Channel=3
+						 * No used
+						 */
+#define SPDIF_SELECT2           0x72            /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
+#define ROUTING2_FRONT_MASK     0x00010000      /* Enable for Front speakers. */
+#define ROUTING2_CENTER_LFE_MASK 0x00020000     /* Enable for Center/LFE speakers. */
+#define ROUTING2_REAR_MASK      0x00080000      /* Enable for Rear speakers. */
+						/* Audio output control
+						 * AC97 output enable [5:0]
+						 * I2S output enable [19:16]
+						 * SPDIF output enable [27:24]
+						 */ 
+#define UNKNOWN73               0x73            /* Unknown. Readonly. Default 0x0 */
+#define CHIP_VERSION            0x74            /* P17 Chip version. Channel_id 0 only. Default 00000071 */
+#define EXTENDED_INT_MASK       0x75            /* Used by both playback and capture interrupt handler */
+						/* Sets which Interrupts are enabled. */
+						/* 0x00000001 = Half period. Playback.
+						 * 0x00000010 = Full period. Playback.
+						 * 0x00000100 = Half buffer. Playback.
+						 * 0x00001000 = Full buffer. Playback.
+						 * 0x00010000 = Half buffer. Capture.
+						 * 0x00100000 = Full buffer. Capture.
+						 * Capture can only do 2 periods.
+						 * 0x01000000 = End audio. Playback.
+						 * 0x40000000 = Half buffer Playback,Caputre xrun.
+						 * 0x80000000 = Full buffer Playback,Caputre xrun.
+						 */
+#define EXTENDED_INT            0x76            /* Used by both playback and capture interrupt handler */
+						/* Shows which interrupts are active at the moment. */
+						/* Same bit layout as EXTENDED_INT_MASK */
+#define COUNTER77               0x77		/* Counter range 0 to 0x3fffff, 192000 counts per second. */
+#define COUNTER78               0x78		/* Counter range 0 to 0x3fffff, 44100 counts per second. */
+#define EXTENDED_INT_TIMER      0x79            /* Channel_id 0 only. Used by both playback and capture interrupt handler */
+						/* Causes interrupts based on timer intervals. */
+#define SPI			0x7a		/* SPI: Serial Interface Register */
+#define I2C_A			0x7b		/* I2C Address. 32 bit */
+#define I2C_D0			0x7c		/* I2C Data Port 0. 32 bit */
+#define I2C_D1			0x7d		/* I2C Data Port 1. 32 bit */
+//I2C values
+#define I2C_A_ADC_ADD_MASK	0x000000fe	//The address is a 7 bit address
+#define I2C_A_ADC_RW_MASK	0x00000001	//bit mask for R/W
+#define I2C_A_ADC_TRANS_MASK	0x00000010  	//Bit mask for I2c address DAC value
+#define I2C_A_ADC_ABORT_MASK	0x00000020	//Bit mask for I2C transaction abort flag
+#define I2C_A_ADC_LAST_MASK	0x00000040	//Bit mask for Last word transaction
+#define I2C_A_ADC_BYTE_MASK	0x00000080	//Bit mask for Byte Mode
+
+#define I2C_A_ADC_ADD		0x00000034	//This is the Device address for ADC 
+#define I2C_A_ADC_READ		0x00000001	//To perform a read operation
+#define I2C_A_ADC_START		0x00000100	//Start I2C transaction
+#define I2C_A_ADC_ABORT		0x00000200	//I2C transaction abort
+#define I2C_A_ADC_LAST		0x00000400	//I2C last transaction
+#define I2C_A_ADC_BYTE		0x00000800	//I2C one byte mode
+
+#define I2C_D_ADC_REG_MASK	0xfe000000  	//ADC address register 
+#define I2C_D_ADC_DAT_MASK	0x01ff0000  	//ADC data register
+
+#define ADC_TIMEOUT		0x00000007	//ADC Timeout Clock Disable
+#define ADC_IFC_CTRL		0x0000000b	//ADC Interface Control
+#define ADC_MASTER		0x0000000c	//ADC Master Mode Control
+#define ADC_POWER		0x0000000d	//ADC PowerDown Control
+#define ADC_ATTEN_ADCL		0x0000000e	//ADC Attenuation ADCL
+#define ADC_ATTEN_ADCR		0x0000000f	//ADC Attenuation ADCR
+#define ADC_ALC_CTRL1		0x00000010	//ADC ALC Control 1
+#define ADC_ALC_CTRL2		0x00000011	//ADC ALC Control 2
+#define ADC_ALC_CTRL3		0x00000012	//ADC ALC Control 3
+#define ADC_NOISE_CTRL		0x00000013	//ADC Noise Gate Control
+#define ADC_LIMIT_CTRL		0x00000014	//ADC Limiter Control
+#define ADC_MUX			0x00000015  	//ADC Mux offset
+
+#if 0
+/* FIXME: Not tested yet. */
+#define ADC_GAIN_MASK		0x000000ff	//Mask for ADC Gain
+#define ADC_ZERODB		0x000000cf	//Value to set ADC to 0dB
+#define ADC_MUTE_MASK		0x000000c0	//Mask for ADC mute
+#define ADC_MUTE		0x000000c0	//Value to mute ADC
+#define ADC_OSR			0x00000008	//Mask for ADC oversample rate select
+#define ADC_TIMEOUT_DISABLE	0x00000008	//Value and mask to disable Timeout clock
+#define ADC_HPF_DISABLE		0x00000100	//Value and mask to disable High pass filter
+#define ADC_TRANWIN_MASK	0x00000070	//Mask for Length of Transient Window
+#endif
+
+#define ADC_MUX_MASK		0x0000000f	//Mask for ADC Mux
+#define ADC_MUX_PHONE		0x00000001	//Value to select TAD at ADC Mux (Not used)
+#define ADC_MUX_MIC		0x00000002	//Value to select Mic at ADC Mux
+#define ADC_MUX_LINEIN		0x00000004	//Value to select LineIn at ADC Mux
+#define ADC_MUX_AUX		0x00000008	//Value to select Aux at ADC Mux
+
+#define SET_CHANNEL 0  /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
+#define PCM_FRONT_CHANNEL 0
+#define PCM_REAR_CHANNEL 1
+#define PCM_CENTER_LFE_CHANNEL 2
+#define PCM_UNKNOWN_CHANNEL 3
+#define CONTROL_FRONT_CHANNEL 0
+#define CONTROL_REAR_CHANNEL 3
+#define CONTROL_CENTER_LFE_CHANNEL 1
+#define CONTROL_UNKNOWN_CHANNEL 2
+
+
+/* Based on WM8768 Datasheet Rev 4.2 page 32 */
+#define SPI_REG_MASK	0x1ff	/* 16-bit SPI writes have a 7-bit address */
+#define SPI_REG_SHIFT	9	/* followed by 9 bits of data */
+
+#define SPI_LDA1_REG		0	/* digital attenuation */
+#define SPI_RDA1_REG		1
+#define SPI_LDA2_REG		4
+#define SPI_RDA2_REG		5
+#define SPI_LDA3_REG		6
+#define SPI_RDA3_REG		7
+#define SPI_LDA4_REG		13
+#define SPI_RDA4_REG		14
+#define SPI_MASTDA_REG		8
+
+#define SPI_DA_BIT_UPDATE	(1<<8)	/* update attenuation values */
+#define SPI_DA_BIT_0dB		0xff	/* 0 dB */
+#define SPI_DA_BIT_infdB	0x00	/* inf dB attenuation (mute) */
+
+#define SPI_PL_REG		2
+#define SPI_PL_BIT_L_M		(0<<5)	/* left channel = mute */
+#define SPI_PL_BIT_L_L		(1<<5)	/* left channel = left */
+#define SPI_PL_BIT_L_R		(2<<5)	/* left channel = right */
+#define SPI_PL_BIT_L_C		(3<<5)	/* left channel = (L+R)/2 */
+#define SPI_PL_BIT_R_M		(0<<7)	/* right channel = mute */
+#define SPI_PL_BIT_R_L		(1<<7)	/* right channel = left */
+#define SPI_PL_BIT_R_R		(2<<7)	/* right channel = right */
+#define SPI_PL_BIT_R_C		(3<<7)	/* right channel = (L+R)/2 */
+#define SPI_IZD_REG		2
+#define SPI_IZD_BIT		(0<<4)	/* infinite zero detect */
+
+#define SPI_FMT_REG		3
+#define SPI_FMT_BIT_RJ		(0<<0)	/* right justified mode */
+#define SPI_FMT_BIT_LJ		(1<<0)	/* left justified mode */
+#define SPI_FMT_BIT_I2S		(2<<0)	/* I2S mode */
+#define SPI_FMT_BIT_DSP		(3<<0)	/* DSP Modes A or B */
+#define SPI_LRP_REG		3
+#define SPI_LRP_BIT		(1<<2)	/* invert LRCLK polarity */
+#define SPI_BCP_REG		3
+#define SPI_BCP_BIT		(1<<3)	/* invert BCLK polarity */
+#define SPI_IWL_REG		3
+#define SPI_IWL_BIT_16		(0<<4)	/* 16-bit world length */
+#define SPI_IWL_BIT_20		(1<<4)	/* 20-bit world length */
+#define SPI_IWL_BIT_24		(2<<4)	/* 24-bit world length */
+#define SPI_IWL_BIT_32		(3<<4)	/* 32-bit world length */
+
+#define SPI_MS_REG		10
+#define SPI_MS_BIT		(1<<5)	/* master mode */
+#define SPI_RATE_REG		10	/* only applies in master mode */
+#define SPI_RATE_BIT_128	(0<<6)	/* MCLK = LRCLK * 128 */
+#define SPI_RATE_BIT_192	(1<<6)
+#define SPI_RATE_BIT_256	(2<<6)
+#define SPI_RATE_BIT_384	(3<<6)
+#define SPI_RATE_BIT_512	(4<<6)
+#define SPI_RATE_BIT_768	(5<<6)
+
+/* They really do label the bit for the 4th channel "4" and not "3" */
+#define SPI_DMUTE0_REG		9
+#define SPI_DMUTE1_REG		9
+#define SPI_DMUTE2_REG		9
+#define SPI_DMUTE4_REG		15
+#define SPI_DMUTE0_BIT		(1<<3)
+#define SPI_DMUTE1_BIT		(1<<4)
+#define SPI_DMUTE2_BIT		(1<<5)
+#define SPI_DMUTE4_BIT		(1<<2)
+
+#define SPI_PHASE0_REG		3
+#define SPI_PHASE1_REG		3
+#define SPI_PHASE2_REG		3
+#define SPI_PHASE4_REG		15
+#define SPI_PHASE0_BIT		(1<<6)
+#define SPI_PHASE1_BIT		(1<<7)
+#define SPI_PHASE2_BIT		(1<<8)
+#define SPI_PHASE4_BIT		(1<<3)
+
+#define SPI_PDWN_REG		2	/* power down all DACs */
+#define SPI_PDWN_BIT		(1<<2)
+#define SPI_DACD0_REG		10	/* power down individual DACs */
+#define SPI_DACD1_REG		10
+#define SPI_DACD2_REG		10
+#define SPI_DACD4_REG		15
+#define SPI_DACD0_BIT		(1<<1)
+#define SPI_DACD1_BIT		(1<<2)
+#define SPI_DACD2_BIT		(1<<3)
+#define SPI_DACD4_BIT		(1<<0)	/* datasheet error says it's 1 */
+
+#define SPI_PWRDNALL_REG	10	/* power down everything */
+#define SPI_PWRDNALL_BIT	(1<<4)
+
+#include "ca_midi.h"
+
+struct snd_ca0106;
+
+struct snd_ca0106_channel {
+	struct snd_ca0106 *emu;
+	int number;
+	int use;
+	void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
+	struct snd_ca0106_pcm *epcm;
+};
+
+struct snd_ca0106_pcm {
+	struct snd_ca0106 *emu;
+	struct snd_pcm_substream *substream;
+        int channel_id;
+	unsigned short running;
+};
+
+struct snd_ca0106_details {
+        u32 serial;
+        char * name;
+	int ac97;	/* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
+			   ac97 = 1 -> Default to AC97 in. */
+	int gpio_type;	/* gpio_type = 1 -> shared mic-in/line-in
+			   gpio_type = 2 -> shared side-out/line-in. */
+	int i2c_adc;	/* with i2c_adc=1, the driver adds some capture volume
+			   controls, phone, mic, line-in and aux. */
+	u16 spi_dac;	/* spi_dac = 0 -> no spi interface for DACs
+			   spi_dac = 0x<front><rear><center-lfe><side>
+			   -> specifies DAC id for each channel pair. */
+};
+
+// definition of the chip-specific record
+struct snd_ca0106 {
+	struct snd_card *card;
+	const struct snd_ca0106_details *details;
+	struct pci_dev *pci;
+
+	unsigned long port;
+	int irq;
+
+	unsigned int serial;            /* serial number */
+	unsigned short model;		/* subsystem id */
+
+	spinlock_t emu_lock;
+
+	struct snd_ac97 *ac97;
+	struct snd_pcm *pcm[4];
+
+	struct snd_ca0106_channel playback_channels[4];
+	struct snd_ca0106_channel capture_channels[4];
+	u32 spdif_bits[4];             /* s/pdif out default setup */
+	u32 spdif_str_bits[4];         /* s/pdif out per-stream setup */
+	int spdif_enable;
+	int capture_source;
+	int i2c_capture_source;
+	u8 i2c_capture_volume[4][2];
+	int capture_mic_line_in;
+
+	struct snd_dma_buffer *buffer;
+
+	struct snd_ca_midi midi;
+	struct snd_ca_midi midi2;
+
+	u16 spi_dac_reg[16];
+
+#ifdef CONFIG_PM_SLEEP
+#define NUM_SAVED_VOLUMES	9
+	unsigned int saved_vol[NUM_SAVED_VOLUMES];
+#endif
+};
+
+int snd_ca0106_mixer(struct snd_ca0106 *emu);
+int snd_ca0106_proc_init(struct snd_ca0106 * emu);
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, 
+				 unsigned int reg, 
+				 unsigned int chn);
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu, 
+			  unsigned int reg, 
+			  unsigned int chn, 
+			  unsigned int data);
+
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+				   unsigned int data);
+
+#ifdef CONFIG_PM_SLEEP
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
+void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
+#else
+#define snd_ca0106_mixer_suspend(chip)	do { } while (0)
+#define snd_ca0106_mixer_resume(chip)	do { } while (0)
+#endif
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
new file mode 100644
index 0000000000..cf1bac7a43
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -0,0 +1,1853 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/*
+ *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ *  Version: 0.0.25
+ *
+ *  FEATURES currently supported:
+ *    Front, Rear and Center/LFE.
+ *    Surround40 and Surround51.
+ *    Capture from MIC an LINE IN input.
+ *    SPDIF digital playback of PCM stereo and AC3/DTS works.
+ *    (One can use a standard mono mini-jack to one RCA plugs cable.
+ *     or one can use a standard stereo mini-jack to two RCA plugs cable.
+ *     Plug one of the RCA plugs into the Coax input of the external decoder/receiver.)
+ *    ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. )
+ *    Notes on how to capture sound:
+ *      The AC97 is used in the PLAYBACK direction.
+ *      The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC.
+ *      So, to record from the MIC, set the MIC Playback volume to max,
+ *      unmute the MIC and turn up the MASTER Playback volume.
+ *      So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume.
+ *   
+ *    The only playback controls that currently do anything are: -
+ *    Analog Front
+ *    Analog Rear
+ *    Analog Center/LFE
+ *    SPDIF Front
+ *    SPDIF Rear
+ *    SPDIF Center/LFE
+ *   
+ *    For capture from Mic in or Line in.
+ *    Digital/Analog ( switch must be in Analog mode for CAPTURE. )
+ * 
+ *    CAPTURE feedback into PLAYBACK
+ * 
+ *  Changelog:
+ *    Support interrupts per period.
+ *    Removed noise from Center/LFE channel when in Analog mode.
+ *    Rename and remove mixer controls.
+ *  0.0.6
+ *    Use separate card based DMA buffer for periods table list.
+ *  0.0.7
+ *    Change remove and rename ctrls into lists.
+ *  0.0.8
+ *    Try to fix capture sources.
+ *  0.0.9
+ *    Fix AC3 output.
+ *    Enable S32_LE format support.
+ *  0.0.10
+ *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ *  0.0.11
+ *    Add Model name recognition.
+ *  0.0.12
+ *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ *    Remove redundent "voice" handling.
+ *  0.0.13
+ *    Single trigger call for multi channels.
+ *  0.0.14
+ *    Set limits based on what the sound card hardware can do.
+ *    playback periods_min=2, periods_max=8
+ *    capture hw constraints require period_size = n * 64 bytes.
+ *    playback hw constraints require period_size = n * 64 bytes.
+ *  0.0.15
+ *    Minor updates.
+ *  0.0.16
+ *    Implement 192000 sample rate.
+ *  0.0.17
+ *    Add support for SB0410 and SB0413.
+ *  0.0.18
+ *    Modified Copyright message.
+ *  0.0.19
+ *    Finally fix support for SB Live 24 bit. SB0410 and SB0413.
+ *    The output codec needs resetting, otherwise all output is muted.
+ *  0.0.20
+ *    Merge "pci_disable_device(pci);" fixes.
+ *  0.0.21
+ *    Add 4 capture channels. (SPDIF only comes in on channel 0. )
+ *    Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.)
+ *  0.0.22
+ *    Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
+ *  0.0.23
+ *    Implement support for Line-in capture on SB Live 24bit.
+ *  0.0.24
+ *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *  0.0.25
+ *    Powerdown SPI DAC channels when not in use
+ *
+ *  BUGS:
+ *    Some stability problems when unloading the snd-ca0106 kernel module.
+ *    --
+ *
+ *  TODO:
+ *    4 Capture channels, only one implemented so far.
+ *    Other capture rates apart from 48khz not implemented.
+ *    MIDI
+ *    --
+ *  GENERAL INFO:
+ *    Model: SB0310
+ *    P17 Chip: CA0106-DAT
+ *    AC97 Codec: STAC 9721
+ *    ADC: Philips 1361T (Stereo 24bit)
+ *    DAC: WM8746EDS (6-channel, 24bit, 192Khz)
+ *
+ *  GENERAL INFO:
+ *    Model: SB0410
+ *    P17 Chip: CA0106-DAT
+ *    AC97 Codec: None
+ *    ADC: WM8775EDS (4 Channel)
+ *    DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support)
+ *    SPDIF Out control switches between Mic in and SPDIF out.
+ *    No sound out or mic input working yet.
+ * 
+ *  GENERAL INFO:
+ *    Model: SB0413
+ *    P17 Chip: CA0106-DAT
+ *    AC97 Codec: None.
+ *    ADC: Unknown
+ *    DAC: Unknown
+ *    Trying to handle it like the SB0410.
+ *
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
+ *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+
+MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>");
+MODULE_DESCRIPTION("CA0106");
+MODULE_LICENSE("GPL");
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard.");
+module_param_array(subsystem, uint, NULL, 0444);
+MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+
+#include "ca0106.h"
+
+static const struct snd_ca0106_details ca0106_chip_details[] = {
+	 /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+	 /* It is really just a normal SB Live 24bit. */
+	 /* Tested:
+	  * See ALSA bug#3251
+	  */
+	 { .serial = 0x10131102,
+	   .name   = "X-Fi Extreme Audio [SBxxxx]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1 } ,
+	 /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */
+	 /* It is really just a normal SB Live 24bit. */
+	 /*
+ 	  * CTRL:CA0111-WTLF
+	  * ADC: WM8775SEDS
+	  * DAC: CS4382-KQZ
+	  */
+	 /* Tested:
+	  * Playback on front, rear, center/lfe speakers
+	  * Capture from Mic in.
+	  * Not-Tested:
+	  * Capture from Line in.
+	  * Playback to digital out.
+	  */
+	 { .serial = 0x10121102,
+	   .name   = "X-Fi Extreme Audio [SB0790]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1 } ,
+	 /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97.  */
+	 /* AudigyLS[SB0310] */
+	 { .serial = 0x10021102,
+	   .name   = "AudigyLS [SB0310]",
+	   .ac97   = 1 } , 
+	 /* Unknown AudigyLS that also says SB0310 on it */
+	 { .serial = 0x10051102,
+	   .name   = "AudigyLS [SB0310b]",
+	   .ac97   = 1 } ,
+	 /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
+	 { .serial = 0x10061102,
+	   .name   = "Live! 7.1 24bit [SB0410]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1 } ,
+	 /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97.  */
+	 { .serial = 0x10071102,
+	   .name   = "Live! 7.1 24bit [SB0413]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1 } ,
+	 /* New Audigy SE. Has a different DAC. */
+	 /* SB0570:
+	  * CTRL:CA0106-DAT
+	  * ADC: WM8775EDS
+	  * DAC: WM8768GEDS
+	  */
+	 { .serial = 0x100a1102,
+	   .name   = "Audigy SE [SB0570]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1,
+	   .spi_dac = 0x4021 } ,
+	 /* New Audigy LS. Has a different DAC. */
+	 /* SB0570:
+	  * CTRL:CA0106-DAT
+	  * ADC: WM8775EDS
+	  * DAC: WM8768GEDS
+	  */
+	 { .serial = 0x10111102,
+	   .name   = "Audigy SE OEM [SB0570a]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1,
+	   .spi_dac = 0x4021 } ,
+	/* Sound Blaster 5.1vx
+	 * Tested: Playback on front, rear, center/lfe speakers
+	 * Not-Tested: Capture
+	 */
+	{ .serial = 0x10041102,
+	  .name   = "Sound Blaster 5.1vx [SB1070]",
+	  .gpio_type = 1,
+	  .i2c_adc = 0,
+	  .spi_dac = 0x0124
+	 } ,
+	 /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+	 /* SB0438
+	  * CTRL:CA0106-DAT
+	  * ADC: WM8775SEDS
+	  * DAC: CS4382-KQZ
+	  */
+	 { .serial = 0x10091462,
+	   .name   = "MSI K8N Diamond MB [SB0438]",
+	   .gpio_type = 2,
+	   .i2c_adc = 1 } ,
+	 /* MSI K8N Diamond PLUS MB */
+	 { .serial = 0x10091102,
+	   .name   = "MSI K8N Diamond MB",
+	   .gpio_type = 2,
+	   .i2c_adc = 1,
+	   .spi_dac = 0x4021 } ,
+	/* Giga-byte GA-G1975X mobo
+	 * Novell bnc#395807
+	 */
+	/* FIXME: the GPIO and I2C setting aren't tested well */
+	{ .serial = 0x1458a006,
+	  .name = "Giga-byte GA-G1975X",
+	  .gpio_type = 1,
+	  .i2c_adc = 1 },
+	 /* Shuttle XPC SD31P which has an onboard Creative Labs
+	  * Sound Blaster Live! 24-bit EAX
+	  * high-definition 7.1 audio processor".
+	  * Added using info from andrewvegan in alsa bug #1298
+	  */
+	 { .serial = 0x30381297,
+	   .name   = "Shuttle XPC SD31P [SD31P]",
+	   .gpio_type = 1,
+	   .i2c_adc = 1 } ,
+	/* Shuttle XPC SD11G5 which has an onboard Creative Labs
+	 * Sound Blaster Live! 24-bit EAX
+	 * high-definition 7.1 audio processor".
+	 * Fixes ALSA bug#1600
+         */
+	{ .serial = 0x30411297,
+	  .name = "Shuttle XPC SD11G5 [SD11G5]",
+	  .gpio_type = 1,
+	  .i2c_adc = 1 } ,
+	 { .serial = 0,
+	   .name   = "AudigyLS [Unknown]" }
+};
+
+/* hardware definition */
+static const struct snd_pcm_hardware snd_ca0106_playback_hw = {
+	.info =			SNDRV_PCM_INFO_MMAP | 
+				SNDRV_PCM_INFO_INTERLEAVED |
+				SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				SNDRV_PCM_INFO_MMAP_VALID |
+				SNDRV_PCM_INFO_SYNC_START,
+	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	.rates =		(SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+				 SNDRV_PCM_RATE_192000),
+	.rate_min =		48000,
+	.rate_max =		192000,
+	.channels_min =		2,  //1,
+	.channels_max =		2,  //6,
+	.buffer_bytes_max =	((65536 - 64) * 8),
+	.period_bytes_min =	64,
+	.period_bytes_max =	(65536 - 64),
+	.periods_min =		2,
+	.periods_max =		8,
+	.fifo_size =		0,
+};
+
+static const struct snd_pcm_hardware snd_ca0106_capture_hw = {
+	.info =			(SNDRV_PCM_INFO_MMAP | 
+				 SNDRV_PCM_INFO_INTERLEAVED |
+				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				 SNDRV_PCM_INFO_MMAP_VALID),
+	.formats =		SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */
+	.rates =		(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+	.rate_min =		44100,
+#else
+	.rates =		(SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000),
+	.rate_min =		48000,
+#endif /* FIXME */
+	.rate_max =		192000,
+	.channels_min =		2,
+	.channels_max =		2,
+	.buffer_bytes_max =	65536 - 128,
+	.period_bytes_min =	64,
+	.period_bytes_max =	32768 - 64,
+	.periods_min =		2,
+	.periods_max =		2,
+	.fifo_size =		0,
+};
+
+unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, 
+					  unsigned int reg, 
+					  unsigned int chn)
+{
+	unsigned long flags;
+	unsigned int regptr, val;
+  
+	regptr = (reg << 16) | chn;
+
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	outl(regptr, emu->port + CA0106_PTR);
+	val = inl(emu->port + CA0106_DATA);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+	return val;
+}
+
+void snd_ca0106_ptr_write(struct snd_ca0106 *emu, 
+				   unsigned int reg, 
+				   unsigned int chn, 
+				   unsigned int data)
+{
+	unsigned int regptr;
+	unsigned long flags;
+
+	regptr = (reg << 16) | chn;
+
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	outl(regptr, emu->port + CA0106_PTR);
+	outl(data, emu->port + CA0106_DATA);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+int snd_ca0106_spi_write(struct snd_ca0106 * emu,
+				   unsigned int data)
+{
+	unsigned int reset, set;
+	unsigned int reg, tmp;
+	int n, result;
+	reg = SPI;
+	if (data > 0xffff) /* Only 16bit values allowed */
+		return 1;
+	tmp = snd_ca0106_ptr_read(emu, reg, 0);
+	reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */
+	set = reset | 0x10000; /* Set xxx1xxxx */
+	snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+	tmp = snd_ca0106_ptr_read(emu, reg, 0); /* write post */
+	snd_ca0106_ptr_write(emu, reg, 0, set | data);
+	result = 1;
+	/* Wait for status bit to return to 0 */
+	for (n = 0; n < 100; n++) {
+		udelay(10);
+		tmp = snd_ca0106_ptr_read(emu, reg, 0);
+		if (!(tmp & 0x10000)) {
+			result = 0;
+			break;
+		}
+	}
+	if (result) /* Timed out */
+		return 1;
+	snd_ca0106_ptr_write(emu, reg, 0, reset | data);
+	tmp = snd_ca0106_ptr_read(emu, reg, 0); /* Write post */
+	return 0;
+}
+
+/* The ADC does not support i2c read, so only write is implemented */
+int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
+				u32 reg,
+				u32 value)
+{
+	u32 tmp;
+	int timeout = 0;
+	int status;
+	int retry;
+	if ((reg > 0x7f) || (value > 0x1ff)) {
+		dev_err(emu->card->dev, "i2c_write: invalid values.\n");
+		return -EINVAL;
+	}
+
+	tmp = reg << 25 | value << 16;
+	/*
+	dev_dbg(emu->card->dev, "I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	*/
+	/* Not sure what this I2C channel controls. */
+	/* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
+
+	/* This controls the I2C connected to the WM8775 ADC Codec */
+	snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp);
+
+	for (retry = 0; retry < 10; retry++) {
+		/* Send the data to i2c */
+		//tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
+		//tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+		tmp = 0;
+		tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
+		snd_ca0106_ptr_write(emu, I2C_A, 0, tmp);
+
+		/* Wait till the transaction ends */
+		while (1) {
+			status = snd_ca0106_ptr_read(emu, I2C_A, 0);
+			/*dev_dbg(emu->card->dev, "I2C:status=0x%x\n", status);*/
+			timeout++;
+			if ((status & I2C_A_ADC_START) == 0)
+				break;
+
+			if (timeout > 1000)
+				break;
+		}
+		//Read back and see if the transaction is successful
+		if ((status & I2C_A_ADC_ABORT) == 0)
+			break;
+	}
+
+	if (retry == 10) {
+		dev_err(emu->card->dev, "Writing to ADC failed!\n");
+		return -EINVAL;
+	}
+    
+    	return 0;
+}
+
+
+static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+	unsigned long flags;
+	unsigned int intr_enable;
+
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	intr_enable = inl(emu->port + CA0106_INTE) | intrenb;
+	outl(intr_enable, emu->port + CA0106_INTE);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb)
+{
+	unsigned long flags;
+	unsigned int intr_enable;
+
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	intr_enable = inl(emu->port + CA0106_INTE) & ~intrenb;
+	outl(intr_enable, emu->port + CA0106_INTE);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+
+static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+	kfree(runtime->private_data);
+}
+
+static const int spi_dacd_reg[] = {
+	SPI_DACD0_REG,
+	SPI_DACD1_REG,
+	SPI_DACD2_REG,
+	0,
+	SPI_DACD4_REG,
+};
+static const int spi_dacd_bit[] = {
+	SPI_DACD0_BIT,
+	SPI_DACD1_BIT,
+	SPI_DACD2_BIT,
+	0,
+	SPI_DACD4_BIT,
+};
+
+static void restore_spdif_bits(struct snd_ca0106 *chip, int idx)
+{
+	if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) {
+		chip->spdif_str_bits[idx] = chip->spdif_bits[idx];
+		snd_ca0106_ptr_write(chip, SPCS0 + idx, 0,
+				     chip->spdif_str_bits[idx]);
+	}
+}
+
+static int snd_ca0106_channel_dac(struct snd_ca0106 *chip,
+				  const struct snd_ca0106_details *details,
+				  int channel_id)
+{
+	switch (channel_id) {
+	case PCM_FRONT_CHANNEL:
+		return (details->spi_dac & 0xf000) >> (4 * 3);
+	case PCM_REAR_CHANNEL:
+		return (details->spi_dac & 0x0f00) >> (4 * 2);
+	case PCM_CENTER_LFE_CHANNEL:
+		return (details->spi_dac & 0x00f0) >> (4 * 1);
+	case PCM_UNKNOWN_CHANNEL:
+		return (details->spi_dac & 0x000f) >> (4 * 0);
+	default:
+		dev_dbg(chip->card->dev, "ca0106: unknown channel_id %d\n",
+			   channel_id);
+	}
+	return 0;
+}
+
+static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id,
+				    int power)
+{
+	if (chip->details->spi_dac) {
+		const int dac = snd_ca0106_channel_dac(chip, chip->details,
+						       channel_id);
+		const int reg = spi_dacd_reg[dac];
+		const int bit = spi_dacd_bit[dac];
+
+		if (power)
+			/* Power up */
+			chip->spi_dac_reg[reg] &= ~bit;
+		else
+			/* Power down */
+			chip->spi_dac_reg[reg] |= bit;
+		if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+			return -ENXIO;
+	}
+	return 0;
+}
+
+/* open_playback callback */
+static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream,
+						int channel_id)
+{
+	struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+        struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]);
+	struct snd_ca0106_pcm *epcm;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+
+	if (epcm == NULL)
+		return -ENOMEM;
+	epcm->emu = chip;
+	epcm->substream = substream;
+        epcm->channel_id=channel_id;
+  
+	runtime->private_data = epcm;
+	runtime->private_free = snd_ca0106_pcm_free_substream;
+  
+	runtime->hw = snd_ca0106_playback_hw;
+
+        channel->emu = chip;
+        channel->number = channel_id;
+
+	channel->use = 1;
+	/*
+	dev_dbg(chip->card->dev, "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
+        //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+	channel->epcm = epcm;
+	err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0)
+                return err;
+	err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+	if (err < 0)
+                return err;
+	snd_pcm_set_sync(substream);
+
+	/* Front channel dac should already be on */
+	if (channel_id != PCM_FRONT_CHANNEL) {
+		err = snd_ca0106_pcm_power_dac(chip, channel_id, 1);
+		if (err < 0)
+			return err;
+	}
+
+	restore_spdif_bits(chip, channel_id);
+
+	return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+        struct snd_ca0106_pcm *epcm = runtime->private_data;
+	chip->playback_channels[epcm->channel_id].use = 0;
+
+	restore_spdif_bits(chip, epcm->channel_id);
+
+	/* Front channel dac should stay on */
+	if (epcm->channel_id != PCM_FRONT_CHANNEL) {
+		int err;
+		err = snd_ca0106_pcm_power_dac(chip, epcm->channel_id, 0);
+		if (err < 0)
+			return err;
+	}
+
+	/* FIXME: maybe zero others */
+	return 0;
+}
+
+static int snd_ca0106_pcm_open_playback_front(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_center_lfe(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_unknown(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_rear(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
+}
+
+/* open_capture callback */
+static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substream,
+					       int channel_id)
+{
+	struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+        struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]);
+	struct snd_ca0106_pcm *epcm;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	int err;
+
+	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
+	if (!epcm)
+		return -ENOMEM;
+
+	epcm->emu = chip;
+	epcm->substream = substream;
+        epcm->channel_id=channel_id;
+  
+	runtime->private_data = epcm;
+	runtime->private_free = snd_ca0106_pcm_free_substream;
+  
+	runtime->hw = snd_ca0106_capture_hw;
+
+        channel->emu = chip;
+        channel->number = channel_id;
+
+	channel->use = 1;
+	/*
+	dev_dbg(chip->card->dev, "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
+        //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+        channel->epcm = epcm;
+	err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0)
+                return err;
+	//snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
+	err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+	if (err < 0)
+                return err;
+	return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_capture(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+        struct snd_ca0106_pcm *epcm = runtime->private_data;
+	chip->capture_channels[epcm->channel_id].use = 0;
+	/* FIXME: maybe zero others */
+	return 0;
+}
+
+static int snd_ca0106_pcm_open_0_capture(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_capture_channel(substream, 0);
+}
+
+static int snd_ca0106_pcm_open_1_capture(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_capture_channel(substream, 1);
+}
+
+static int snd_ca0106_pcm_open_2_capture(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_capture_channel(substream, 2);
+}
+
+static int snd_ca0106_pcm_open_3_capture(struct snd_pcm_substream *substream)
+{
+	return snd_ca0106_pcm_open_capture_channel(substream, 3);
+}
+
+/* prepare playback callback */
+static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_ca0106_pcm *epcm = runtime->private_data;
+	int channel = epcm->channel_id;
+	u32 *table_base = (u32 *)(emu->buffer->area+(8*16*channel));
+	u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+	u32 hcfg_mask = HCFG_PLAYBACK_S32_LE;
+	u32 hcfg_set = 0x00000000;
+	u32 hcfg;
+	u32 reg40_mask = 0x30000 << (channel<<1);
+	u32 reg40_set = 0;
+	u32 reg40;
+	/* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */
+	u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */
+	u32 reg71_set = 0;
+	u32 reg71;
+	int i;
+	
+#if 0 /* debug */
+	dev_dbg(emu->card->dev,
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+	dev_dbg(emu->card->dev,
+		"dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	dev_dbg(emu->card->dev,
+		"dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer->addr, emu->buffer->area, emu->buffer->bytes);
+#endif /* debug */
+	/* Rate can be set per channel. */
+	/* reg40 control host to fifo */
+	/* reg71 controls DAC rate. */
+	switch (runtime->rate) {
+	case 44100:
+		reg40_set = 0x10000 << (channel<<1);
+		reg71_set = 0x01010000; 
+		break;
+        case 48000:
+		reg40_set = 0;
+		reg71_set = 0; 
+		break;
+	case 96000:
+		reg40_set = 0x20000 << (channel<<1);
+		reg71_set = 0x02020000; 
+		break;
+	case 192000:
+		reg40_set = 0x30000 << (channel<<1);
+		reg71_set = 0x03030000; 
+		break;
+	default:
+		reg40_set = 0;
+		reg71_set = 0; 
+		break;
+	}
+	/* Format is a global setting */
+	/* FIXME: Only let the first channel accessed set this. */
+	switch (runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		hcfg_set = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		hcfg_set = HCFG_PLAYBACK_S32_LE;
+		break;
+	default:
+		hcfg_set = 0;
+		break;
+	}
+	hcfg = inl(emu->port + CA0106_HCFG) ;
+	hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+	outl(hcfg, emu->port + CA0106_HCFG);
+	reg40 = snd_ca0106_ptr_read(emu, 0x40, 0);
+	reg40 = (reg40 & ~reg40_mask) | reg40_set;
+	snd_ca0106_ptr_write(emu, 0x40, 0, reg40);
+	reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+	reg71 = (reg71 & ~reg71_mask) | reg71_set;
+	snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+
+	/* FIXME: Check emu->buffer->size before actually writing to it. */
+        for(i=0; i < runtime->periods; i++) {
+		table_base[i*2] = runtime->dma_addr + (i * period_size_bytes);
+		table_base[i*2+1] = period_size_bytes << 16;
+	}
+ 
+	snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer->addr+(8*16*channel));
+	snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
+	snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0);
+	snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
+	snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+	/* FIXME  test what 0 bytes does. */
+	snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes
+	snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0);
+	snd_ca0106_ptr_write(emu, 0x07, channel, 0x0);
+	snd_ca0106_ptr_write(emu, 0x08, channel, 0);
+        snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
+#if 0
+	snd_ca0106_ptr_write(emu, SPCS0, 0,
+			       SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+			       SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+			       SPCS_GENERATIONSTATUS | 0x00001200 |
+			       0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
+#endif
+
+	return 0;
+}
+
+/* prepare capture callback */
+static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_ca0106_pcm *epcm = runtime->private_data;
+	int channel = epcm->channel_id;
+	u32 hcfg_mask = HCFG_CAPTURE_S32_LE;
+	u32 hcfg_set = 0x00000000;
+	u32 hcfg;
+	u32 over_sampling=0x2;
+	u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */
+	u32 reg71_set = 0;
+	u32 reg71;
+	
+#if 0 /* debug */
+	dev_dbg(emu->card->dev,
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+	dev_dbg(emu->card->dev,
+		"dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	dev_dbg(emu->card->dev,
+		"dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer->addr, emu->buffer->area, emu->buffer->bytes);
+#endif /* debug */
+	/* reg71 controls ADC rate. */
+	switch (runtime->rate) {
+	case 44100:
+		reg71_set = 0x00004000;
+		break;
+        case 48000:
+		reg71_set = 0; 
+		break;
+	case 96000:
+		reg71_set = 0x00008000;
+		over_sampling=0xa;
+		break;
+	case 192000:
+		reg71_set = 0x0000c000; 
+		over_sampling=0xa;
+		break;
+	default:
+		reg71_set = 0; 
+		break;
+	}
+	/* Format is a global setting */
+	/* FIXME: Only let the first channel accessed set this. */
+	switch (runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		hcfg_set = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		hcfg_set = HCFG_CAPTURE_S32_LE;
+		break;
+	default:
+		hcfg_set = 0;
+		break;
+	}
+	hcfg = inl(emu->port + CA0106_HCFG) ;
+	hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+	outl(hcfg, emu->port + CA0106_HCFG);
+	reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+	reg71 = (reg71 & ~reg71_mask) | reg71_set;
+	snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+        if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+	        snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */
+	}
+
+
+	/*
+	dev_dbg(emu->card->dev,
+	       "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, "
+	       "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
+	snd_ca0106_ptr_write(emu, 0x13, channel, 0);
+	snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
+	snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+	snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0);
+
+	return 0;
+}
+
+/* trigger_playback callback */
+static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
+				    int cmd)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime;
+	struct snd_ca0106_pcm *epcm;
+	int channel;
+	int result = 0;
+        struct snd_pcm_substream *s;
+	u32 basic = 0;
+	u32 extended = 0;
+	u32 bits;
+	int running = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		running = 1;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	default:
+		running = 0;
+		break;
+	}
+        snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) != emu ||
+		    s->stream != SNDRV_PCM_STREAM_PLAYBACK)
+			continue;
+		runtime = s->runtime;
+		epcm = runtime->private_data;
+		channel = epcm->channel_id;
+		/* dev_dbg(emu->card->dev, "channel=%d\n", channel); */
+		epcm->running = running;
+		basic |= (0x1 << channel);
+		extended |= (0x10 << channel);
+                snd_pcm_trigger_done(s, substream);
+        }
+	/* dev_dbg(emu->card->dev, "basic=0x%x, extended=0x%x\n",basic, extended); */
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+		bits |= extended;
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
+		bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+		bits |= basic;
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0);
+		bits &= ~basic;
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits);
+		bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0);
+		bits &= ~extended;
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits);
+		break;
+	default:
+		result = -EINVAL;
+		break;
+	}
+	return result;
+}
+
+/* trigger_capture callback */
+static int snd_ca0106_pcm_trigger_capture(struct snd_pcm_substream *substream,
+				    int cmd)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_ca0106_pcm *epcm = runtime->private_data;
+	int channel = epcm->channel_id;
+	int result = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel));
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
+		epcm->running = 1;
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+		snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
+		epcm->running = 0;
+		break;
+	default:
+		result = -EINVAL;
+		break;
+	}
+	return result;
+}
+
+/* pointer_playback callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_ca0106_pcm *epcm = runtime->private_data;
+	unsigned int ptr, prev_ptr;
+	int channel = epcm->channel_id;
+	int timeout = 10;
+
+	if (!epcm->running)
+		return 0;
+
+	prev_ptr = -1;
+	do {
+		ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+		ptr = (ptr >> 3) * runtime->period_size;
+		ptr += bytes_to_frames(runtime,
+			snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel));
+		if (ptr >= runtime->buffer_size)
+			ptr -= runtime->buffer_size;
+		if (prev_ptr == ptr)
+			return ptr;
+		prev_ptr = ptr;
+	} while (--timeout);
+	dev_warn(emu->card->dev, "ca0106: unstable DMA pointer!\n");
+	return 0;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+	struct snd_ca0106 *emu = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_ca0106_pcm *epcm = runtime->private_data;
+	snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+	int channel = epcm->channel_id;
+
+	if (!epcm->running)
+		return 0;
+
+	ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel);
+	ptr2 = bytes_to_frames(runtime, ptr1);
+	ptr=ptr2;
+        if (ptr >= runtime->buffer_size)
+		ptr -= runtime->buffer_size;
+	/*
+	dev_dbg(emu->card->dev, "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
+	return ptr;
+}
+
+/* operators */
+static const struct snd_pcm_ops snd_ca0106_playback_front_ops = {
+	.open =        snd_ca0106_pcm_open_playback_front,
+	.close =       snd_ca0106_pcm_close_playback,
+	.prepare =     snd_ca0106_pcm_prepare_playback,
+	.trigger =     snd_ca0106_pcm_trigger_playback,
+	.pointer =     snd_ca0106_pcm_pointer_playback,
+};
+
+static const struct snd_pcm_ops snd_ca0106_capture_0_ops = {
+	.open =        snd_ca0106_pcm_open_0_capture,
+	.close =       snd_ca0106_pcm_close_capture,
+	.prepare =     snd_ca0106_pcm_prepare_capture,
+	.trigger =     snd_ca0106_pcm_trigger_capture,
+	.pointer =     snd_ca0106_pcm_pointer_capture,
+};
+
+static const struct snd_pcm_ops snd_ca0106_capture_1_ops = {
+	.open =        snd_ca0106_pcm_open_1_capture,
+	.close =       snd_ca0106_pcm_close_capture,
+	.prepare =     snd_ca0106_pcm_prepare_capture,
+	.trigger =     snd_ca0106_pcm_trigger_capture,
+	.pointer =     snd_ca0106_pcm_pointer_capture,
+};
+
+static const struct snd_pcm_ops snd_ca0106_capture_2_ops = {
+	.open =        snd_ca0106_pcm_open_2_capture,
+	.close =       snd_ca0106_pcm_close_capture,
+	.prepare =     snd_ca0106_pcm_prepare_capture,
+	.trigger =     snd_ca0106_pcm_trigger_capture,
+	.pointer =     snd_ca0106_pcm_pointer_capture,
+};
+
+static const struct snd_pcm_ops snd_ca0106_capture_3_ops = {
+	.open =        snd_ca0106_pcm_open_3_capture,
+	.close =       snd_ca0106_pcm_close_capture,
+	.prepare =     snd_ca0106_pcm_prepare_capture,
+	.trigger =     snd_ca0106_pcm_trigger_capture,
+	.pointer =     snd_ca0106_pcm_pointer_capture,
+};
+
+static const struct snd_pcm_ops snd_ca0106_playback_center_lfe_ops = {
+        .open =         snd_ca0106_pcm_open_playback_center_lfe,
+        .close =        snd_ca0106_pcm_close_playback,
+        .prepare =      snd_ca0106_pcm_prepare_playback,     
+        .trigger =      snd_ca0106_pcm_trigger_playback,  
+        .pointer =      snd_ca0106_pcm_pointer_playback, 
+};
+
+static const struct snd_pcm_ops snd_ca0106_playback_unknown_ops = {
+        .open =         snd_ca0106_pcm_open_playback_unknown,
+        .close =        snd_ca0106_pcm_close_playback,
+        .prepare =      snd_ca0106_pcm_prepare_playback,     
+        .trigger =      snd_ca0106_pcm_trigger_playback,  
+        .pointer =      snd_ca0106_pcm_pointer_playback, 
+};
+
+static const struct snd_pcm_ops snd_ca0106_playback_rear_ops = {
+        .open =         snd_ca0106_pcm_open_playback_rear,
+        .close =        snd_ca0106_pcm_close_playback,
+        .prepare =      snd_ca0106_pcm_prepare_playback,     
+        .trigger =      snd_ca0106_pcm_trigger_playback,  
+        .pointer =      snd_ca0106_pcm_pointer_playback, 
+};
+
+
+static unsigned short snd_ca0106_ac97_read(struct snd_ac97 *ac97,
+					     unsigned short reg)
+{
+	struct snd_ca0106 *emu = ac97->private_data;
+	unsigned long flags;
+	unsigned short val;
+
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	outb(reg, emu->port + CA0106_AC97ADDRESS);
+	val = inw(emu->port + CA0106_AC97DATA);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+	return val;
+}
+
+static void snd_ca0106_ac97_write(struct snd_ac97 *ac97,
+				    unsigned short reg, unsigned short val)
+{
+	struct snd_ca0106 *emu = ac97->private_data;
+	unsigned long flags;
+  
+	spin_lock_irqsave(&emu->emu_lock, flags);
+	outb(reg, emu->port + CA0106_AC97ADDRESS);
+	outw(val, emu->port + CA0106_AC97DATA);
+	spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static int snd_ca0106_ac97(struct snd_ca0106 *chip)
+{
+	struct snd_ac97_bus *pbus;
+	struct snd_ac97_template ac97;
+	int err;
+	static const struct snd_ac97_bus_ops ops = {
+		.write = snd_ca0106_ac97_write,
+		.read = snd_ca0106_ac97_read,
+	};
+  
+	err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus);
+	if (err < 0)
+		return err;
+	pbus->no_vra = 1; /* we don't need VRA */
+
+	memset(&ac97, 0, sizeof(ac97));
+	ac97.private_data = chip;
+	ac97.scaps = AC97_SCAP_NO_SPDIF;
+	return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+}
+
+static void ca0106_stop_chip(struct snd_ca0106 *chip);
+
+static void snd_ca0106_free(struct snd_card *card)
+{
+	struct snd_ca0106 *chip = card->private_data;
+
+	ca0106_stop_chip(chip);
+}
+
+static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
+{
+	unsigned int status;
+
+	struct snd_ca0106 *chip = dev_id;
+	int i;
+	int mask;
+        unsigned int stat76;
+	struct snd_ca0106_channel *pchannel;
+
+	status = inl(chip->port + CA0106_IPR);
+	if (! status)
+		return IRQ_NONE;
+
+        stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
+	/*
+	dev_dbg(emu->card->dev, "interrupt status = 0x%08x, stat76=0x%08x\n",
+		   status, stat76);
+	dev_dbg(emu->card->dev, "ptr=0x%08x\n",
+		   snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	*/
+        mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
+	for(i = 0; i < 4; i++) {
+		pchannel = &(chip->playback_channels[i]);
+		if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+			if(pchannel->use) {
+				snd_pcm_period_elapsed(pchannel->epcm->substream);
+				/* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */
+                        }
+		}
+		/*
+		dev_dbg(emu->card->dev, "channel=%p\n", pchannel);
+		dev_dbg(emu->card->dev, "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+		*/
+		mask <<= 1;
+	}
+        mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */
+	for(i = 0; i < 4; i++) {
+		pchannel = &(chip->capture_channels[i]);
+		if (stat76 & mask) {
+/* FIXME: Select the correct substream for period elapsed */
+			if(pchannel->use) {
+				snd_pcm_period_elapsed(pchannel->epcm->substream);
+				/* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */
+                        }
+		}
+		/*
+		dev_dbg(emu->card->dev, "channel=%p\n", pchannel);
+		dev_dbg(emu->card->dev, "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
+		*/
+		mask <<= 1;
+	}
+
+        snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76);
+
+	if (chip->midi.dev_id &&
+	    (status & (chip->midi.ipr_tx|chip->midi.ipr_rx))) {
+		if (chip->midi.interrupt)
+			chip->midi.interrupt(&chip->midi, status);
+		else
+			chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable);
+	}
+
+	// acknowledge the interrupt if necessary
+	outl(status, chip->port + CA0106_IPR);
+
+	return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_chmap_elem surround_map[] = {
+	{ .channels = 2,
+	  .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+	{ }
+};
+
+static const struct snd_pcm_chmap_elem clfe_map[] = {
+	{ .channels = 2,
+	  .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } },
+	{ }
+};
+
+static const struct snd_pcm_chmap_elem side_map[] = {
+	{ .channels = 2,
+	  .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } },
+	{ }
+};
+
+static int snd_ca0106_pcm(struct snd_ca0106 *emu, int device)
+{
+	struct snd_pcm *pcm;
+	struct snd_pcm_substream *substream;
+	const struct snd_pcm_chmap_elem *map = NULL;
+	int err;
+  
+	err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+  
+	pcm->private_data = emu;
+
+	switch (device) {
+	case 0:
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops);
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops);
+	  map = snd_pcm_std_chmaps;
+          break;
+	case 1:
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops);
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops);
+	  map = surround_map;
+          break;
+	case 2:
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops);
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops);
+	  map = clfe_map;
+          break;
+	case 3:
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops);
+	  snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops);
+	  map = side_map;
+          break;
+        }
+
+	pcm->info_flags = 0;
+	strcpy(pcm->name, "CA0106");
+
+	for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; 
+	    substream; 
+	    substream = substream->next) {
+		snd_pcm_set_managed_buffer(substream, SNDRV_DMA_TYPE_DEV,
+					   &emu->pci->dev,
+					   64*1024, 64*1024);
+	}
+
+	for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 
+	      substream; 
+	      substream = substream->next) {
+		snd_pcm_set_managed_buffer(substream, SNDRV_DMA_TYPE_DEV,
+					   &emu->pci->dev,
+					   64*1024, 64*1024);
+	}
+  
+	err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2,
+				     1 << 2, NULL);
+	if (err < 0)
+		return err;
+
+	emu->pcm[device] = pcm;
+  
+	return 0;
+}
+
+#define SPI_REG(reg, value)	(((reg) << SPI_REG_SHIFT) | (value))
+static const unsigned int spi_dac_init[] = {
+	SPI_REG(SPI_LDA1_REG,	SPI_DA_BIT_0dB), /* 0dB dig. attenuation */
+	SPI_REG(SPI_RDA1_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_PL_REG,	SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT),
+	SPI_REG(SPI_FMT_REG,	SPI_FMT_BIT_I2S | SPI_IWL_BIT_24),
+	SPI_REG(SPI_LDA2_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_RDA2_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_LDA3_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_RDA3_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_MASTDA_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(9,		0x00),
+	SPI_REG(SPI_MS_REG,	SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT),
+	SPI_REG(12,		0x00),
+	SPI_REG(SPI_LDA4_REG,	SPI_DA_BIT_0dB),
+	SPI_REG(SPI_RDA4_REG,	SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE),
+	SPI_REG(SPI_DACD4_REG,	SPI_DACD4_BIT),
+};
+
+static const unsigned int i2c_adc_init[][2] = {
+	{ 0x17, 0x00 }, /* Reset */
+	{ 0x07, 0x00 }, /* Timeout */
+	{ 0x0b, 0x22 },  /* Interface control */
+	{ 0x0c, 0x22 },  /* Master mode control */
+	{ 0x0d, 0x08 },  /* Powerdown control */
+	{ 0x0e, 0xcf },  /* Attenuation Left  0x01 = -103dB, 0xff = 24dB */
+	{ 0x0f, 0xcf },  /* Attenuation Right 0.5dB steps */
+	{ 0x10, 0x7b },  /* ALC Control 1 */
+	{ 0x11, 0x00 },  /* ALC Control 2 */
+	{ 0x12, 0x32 },  /* ALC Control 3 */
+	{ 0x13, 0x00 },  /* Noise gate control */
+	{ 0x14, 0xa6 },  /* Limiter control */
+	{ 0x15, ADC_MUX_LINEIN },  /* ADC Mixer control */
+};
+
+static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
+{
+	int ch;
+	unsigned int def_bits;
+
+	outl(0, chip->port + CA0106_INTE);
+
+	/*
+	 *  Init to 0x02109204 :
+	 *  Clock accuracy    = 0     (1000ppm)
+	 *  Sample Rate       = 2     (48kHz)
+	 *  Audio Channel     = 1     (Left of 2)
+	 *  Source Number     = 0     (Unspecified)
+	 *  Generation Status = 1     (Original for Cat Code 12)
+	 *  Cat Code          = 12    (Digital Signal Mixer)
+	 *  Mode              = 0     (Mode 0)
+	 *  Emphasis          = 0     (None)
+	 *  CP                = 1     (Copyright unasserted)
+	 *  AN                = 0     (Audio data)
+	 *  P                 = 0     (Consumer)
+	 */
+	def_bits =
+		SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+		SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+		SPCS_GENERATIONSTATUS | 0x00001200 |
+		0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT;
+	if (!resume) {
+		chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits;
+		chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits;
+		chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits;
+		chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits;
+	}
+	/* Only SPCS1 has been tested */
+	snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]);
+	snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]);
+	snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]);
+	snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]);
+
+        snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
+        snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
+
+        /* Write 0x8000 to AC97_REC_GAIN to mute it. */
+        outb(AC97_REC_GAIN, chip->port + CA0106_AC97ADDRESS);
+        outw(0x8000, chip->port + CA0106_AC97DATA);
+#if 0 /* FIXME: what are these? */
+	snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
+	snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
+	snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
+	snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
+#endif
+
+	/* OSS drivers set this. */
+	/* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */
+
+	/* Analog or Digital output */
+	snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
+	/* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers.
+	 * Use 0x000f0000 for surround71
+	 */
+	snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000);
+
+	chip->spdif_enable = 0; /* Set digital SPDIF output off */
+	/*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */
+	/*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */
+
+	/* goes to 0x40c80000 when doing SPDIF IN/OUT */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000);
+	/* (Mute) CAPTURE feedback into PLAYBACK volume.
+	 * Only lower 16 bits matter.
+	 */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff);
+	/* SPDIF IN Volume */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000);
+	/* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
+	snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000);
+
+	snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
+	snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
+	snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
+	snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
+
+	for (ch = 0; ch < 4; ch++) {
+		/* Only high 16 bits matter */
+		snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030);
+		snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
+#if 0 /* Mute */
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff);
+		snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff);
+#endif
+	}
+	if (chip->details->i2c_adc == 1) {
+	        /* Select MIC, Line in, TAD in, AUX in */
+	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+		/* Default to CAPTURE_SOURCE to i2s in */
+		if (!resume)
+			chip->capture_source = 3;
+	} else if (chip->details->ac97 == 1) {
+	        /* Default to AC97 in */
+	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4);
+		/* Default to CAPTURE_SOURCE to AC97 in */
+		if (!resume)
+			chip->capture_source = 4;
+	} else {
+	        /* Select MIC, Line in, TAD in, AUX in */
+	        snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4);
+		/* Default to Set CAPTURE_SOURCE to i2s in */
+		if (!resume)
+			chip->capture_source = 3;
+	}
+
+	if (chip->details->gpio_type == 2) {
+		/* The SB0438 use GPIO differently. */
+		/* FIXME: Still need to find out what the other GPIO bits do.
+		 * E.g. For digital spdif out.
+		 */
+		outl(0x0, chip->port + CA0106_GPIO);
+		/* outl(0x00f0e000, chip->port + CA0106_GPIO); */ /* Analog */
+		outl(0x005f5301, chip->port + CA0106_GPIO); /* Analog */
+	} else if (chip->details->gpio_type == 1) {
+		/* The SB0410 and SB0413 use GPIO differently. */
+		/* FIXME: Still need to find out what the other GPIO bits do.
+		 * E.g. For digital spdif out.
+		 */
+		outl(0x0, chip->port + CA0106_GPIO);
+		/* outl(0x00f0e000, chip->port + CA0106_GPIO); */ /* Analog */
+		outl(0x005f5301, chip->port + CA0106_GPIO); /* Analog */
+	} else {
+		outl(0x0, chip->port + CA0106_GPIO);
+		outl(0x005f03a3, chip->port + CA0106_GPIO); /* Analog */
+		/* outl(0x005f02a2, chip->port + CA0106_GPIO); */ /* SPDIF */
+	}
+	snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
+
+	/* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */
+	/* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
+	/* outl(0x00001409, chip->port + CA0106_HCFG); */
+	/* outl(0x00000009, chip->port + CA0106_HCFG); */
+	/* AC97 2.0, Enable outputs. */
+	outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port + CA0106_HCFG);
+
+	if (chip->details->i2c_adc == 1) {
+		/* The SB0410 and SB0413 use I2C to control ADC. */
+		int size, n;
+
+		size = ARRAY_SIZE(i2c_adc_init);
+		/* dev_dbg(emu->card->dev, "I2C:array size=0x%x\n", size); */
+		for (n = 0; n < size; n++)
+			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
+					     i2c_adc_init[n][1]);
+		for (n = 0; n < 4; n++) {
+			chip->i2c_capture_volume[n][0] = 0xcf;
+			chip->i2c_capture_volume[n][1] = 0xcf;
+		}
+		chip->i2c_capture_source = 2; /* Line in */
+		/* Enable Line-in capture. MIC in currently untested. */
+		/* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */
+	}
+
+	if (chip->details->spi_dac) {
+		/* The SB0570 use SPI to control DAC. */
+		int size, n;
+
+		size = ARRAY_SIZE(spi_dac_init);
+		for (n = 0; n < size; n++) {
+			int reg = spi_dac_init[n] >> SPI_REG_SHIFT;
+
+			snd_ca0106_spi_write(chip, spi_dac_init[n]);
+			if (reg < ARRAY_SIZE(chip->spi_dac_reg))
+				chip->spi_dac_reg[reg] = spi_dac_init[n];
+		}
+
+		/* Enable front dac only */
+		snd_ca0106_pcm_power_dac(chip, PCM_FRONT_CHANNEL, 1);
+	}
+}
+
+static void ca0106_stop_chip(struct snd_ca0106 *chip)
+{
+	/* disable interrupts */
+	snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+	outl(0, chip->port + CA0106_INTE);
+	snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
+	udelay(1000);
+	/* disable audio */
+	/* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */
+	outl(0, chip->port + CA0106_HCFG);
+	/* FIXME: We need to stop and DMA transfers here.
+	 *        But as I am not sure how yet, we cannot from the dma pages.
+	 * So we can fix: snd-malloc: Memory leak?  pages not freed = 8
+	 */
+}
+
+static int snd_ca0106_create(int dev, struct snd_card *card,
+			     struct pci_dev *pci)
+{
+	struct snd_ca0106 *chip = card->private_data;
+	const struct snd_ca0106_details *c;
+	int err;
+
+	err = pcim_enable_device(pci);
+	if (err < 0)
+		return err;
+	if (dma_set_mask_and_coherent(&pci->dev, DMA_BIT_MASK(32))) {
+		dev_err(card->dev, "error to set 32bit mask DMA\n");
+		return -ENXIO;
+	}
+
+	chip->card = card;
+	chip->pci = pci;
+	chip->irq = -1;
+
+	spin_lock_init(&chip->emu_lock);
+
+	err = pci_request_regions(pci, "snd_ca0106");
+	if (err < 0)
+		return err;
+	chip->port = pci_resource_start(pci, 0);
+
+	if (devm_request_irq(&pci->dev, pci->irq, snd_ca0106_interrupt,
+			     IRQF_SHARED, KBUILD_MODNAME, chip)) {
+		dev_err(card->dev, "cannot grab irq\n");
+		return -EBUSY;
+	}
+	chip->irq = pci->irq;
+	card->sync_irq = chip->irq;
+
+	/* This stores the periods table. */
+	chip->buffer = snd_devm_alloc_pages(&pci->dev, SNDRV_DMA_TYPE_DEV, 1024);
+	if (!chip->buffer)
+		return -ENOMEM;
+
+	pci_set_master(pci);
+	/* read serial */
+	pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+	pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+	dev_info(card->dev, "Model %04x Rev %08x Serial %08x\n",
+	       chip->model, pci->revision, chip->serial);
+	strcpy(card->driver, "CA0106");
+	strcpy(card->shortname, "CA0106");
+
+	for (c = ca0106_chip_details; c->serial; c++) {
+		if (subsystem[dev]) {
+			if (c->serial == subsystem[dev])
+				break;
+		} else if (c->serial == chip->serial)
+			break;
+	}
+	chip->details = c;
+	if (subsystem[dev]) {
+		dev_info(card->dev, "Sound card name=%s, "
+		       "subsystem=0x%x. Forced to subsystem=0x%x\n",
+		       c->name, chip->serial, subsystem[dev]);
+	}
+
+	sprintf(card->longname, "%s at 0x%lx irq %i",
+		c->name, chip->port, chip->irq);
+
+	ca0106_init_chip(chip, 0);
+	return 0;
+}
+
+
+static void ca0106_midi_interrupt_enable(struct snd_ca_midi *midi, int intr)
+{
+	snd_ca0106_intr_enable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static void ca0106_midi_interrupt_disable(struct snd_ca_midi *midi, int intr)
+{
+	snd_ca0106_intr_disable((struct snd_ca0106 *)(midi->dev_id), intr);
+}
+
+static unsigned char ca0106_midi_read(struct snd_ca_midi *midi, int idx)
+{
+	return (unsigned char)snd_ca0106_ptr_read((struct snd_ca0106 *)(midi->dev_id),
+						  midi->port + idx, 0);
+}
+
+static void ca0106_midi_write(struct snd_ca_midi *midi, int data, int idx)
+{
+	snd_ca0106_ptr_write((struct snd_ca0106 *)(midi->dev_id), midi->port + idx, 0, data);
+}
+
+static struct snd_card *ca0106_dev_id_card(void *dev_id)
+{
+	return ((struct snd_ca0106 *)dev_id)->card;
+}
+
+static int ca0106_dev_id_port(void *dev_id)
+{
+	return ((struct snd_ca0106 *)dev_id)->port;
+}
+
+static int snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int channel)
+{
+	struct snd_ca_midi *midi;
+	char *name;
+	int err;
+
+	if (channel == CA0106_MIDI_CHAN_B) {
+		name = "CA0106 MPU-401 (UART) B";
+		midi =  &chip->midi2;
+		midi->tx_enable = INTE_MIDI_TX_B;
+		midi->rx_enable = INTE_MIDI_RX_B;
+		midi->ipr_tx = IPR_MIDI_TX_B;
+		midi->ipr_rx = IPR_MIDI_RX_B;
+		midi->port = MIDI_UART_B_DATA;
+	} else {
+		name = "CA0106 MPU-401 (UART)";
+		midi =  &chip->midi;
+		midi->tx_enable = INTE_MIDI_TX_A;
+		midi->rx_enable = INTE_MIDI_TX_B;
+		midi->ipr_tx = IPR_MIDI_TX_A;
+		midi->ipr_rx = IPR_MIDI_RX_A;
+		midi->port = MIDI_UART_A_DATA;
+	}
+
+	midi->reset = CA0106_MPU401_RESET;
+	midi->enter_uart = CA0106_MPU401_ENTER_UART;
+	midi->ack = CA0106_MPU401_ACK;
+
+	midi->input_avail = CA0106_MIDI_INPUT_AVAIL;
+	midi->output_ready = CA0106_MIDI_OUTPUT_READY;
+
+	midi->channel = channel;
+
+	midi->interrupt_enable = ca0106_midi_interrupt_enable;
+	midi->interrupt_disable = ca0106_midi_interrupt_disable;
+
+	midi->read = ca0106_midi_read;
+	midi->write = ca0106_midi_write;
+
+	midi->get_dev_id_card = ca0106_dev_id_card;
+	midi->get_dev_id_port = ca0106_dev_id_port;
+
+	midi->dev_id = chip;
+	
+	err = ca_midi_init(chip, midi, 0, name);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+
+static int __snd_ca0106_probe(struct pci_dev *pci,
+			      const struct pci_device_id *pci_id)
+{
+	static int dev;
+	struct snd_card *card;
+	struct snd_ca0106 *chip;
+	int i, err;
+
+	if (dev >= SNDRV_CARDS)
+		return -ENODEV;
+	if (!enable[dev]) {
+		dev++;
+		return -ENOENT;
+	}
+
+	err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
+				sizeof(*chip), &card);
+	if (err < 0)
+		return err;
+	chip = card->private_data;
+
+	err = snd_ca0106_create(dev, card, pci);
+	if (err < 0)
+		return err;
+	card->private_free = snd_ca0106_free;
+
+	for (i = 0; i < 4; i++) {
+		err = snd_ca0106_pcm(chip, i);
+		if (err < 0)
+			return err;
+	}
+
+	if (chip->details->ac97 == 1) {
+		/* The SB0410 and SB0413 do not have an AC97 chip. */
+		err = snd_ca0106_ac97(chip);
+		if (err < 0)
+			return err;
+	}
+	err = snd_ca0106_mixer(chip);
+	if (err < 0)
+		return err;
+
+	dev_dbg(card->dev, "probe for MIDI channel A ...");
+	err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A);
+	if (err < 0)
+		return err;
+	dev_dbg(card->dev, " done.\n");
+
+#ifdef CONFIG_SND_PROC_FS
+	snd_ca0106_proc_init(chip);
+#endif
+
+	err = snd_card_register(card);
+	if (err < 0)
+		return err;
+
+	pci_set_drvdata(pci, card);
+	dev++;
+	return 0;
+}
+
+static int snd_ca0106_probe(struct pci_dev *pci,
+			    const struct pci_device_id *pci_id)
+{
+	return snd_card_free_on_error(&pci->dev, __snd_ca0106_probe(pci, pci_id));
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int snd_ca0106_suspend(struct device *dev)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	struct snd_ca0106 *chip = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	if (chip->details->ac97)
+		snd_ac97_suspend(chip->ac97);
+	snd_ca0106_mixer_suspend(chip);
+
+	ca0106_stop_chip(chip);
+	return 0;
+}
+
+static int snd_ca0106_resume(struct device *dev)
+{
+	struct snd_card *card = dev_get_drvdata(dev);
+	struct snd_ca0106 *chip = card->private_data;
+	int i;
+
+	ca0106_init_chip(chip, 1);
+
+	if (chip->details->ac97)
+		snd_ac97_resume(chip->ac97);
+	snd_ca0106_mixer_resume(chip);
+	if (chip->details->spi_dac) {
+		for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++)
+			snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+
+static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume);
+#define SND_CA0106_PM_OPS	&snd_ca0106_pm
+#else
+#define SND_CA0106_PM_OPS	NULL
+#endif
+
+// PCI IDs
+static const struct pci_device_id snd_ca0106_ids[] = {
+	{ PCI_VDEVICE(CREATIVE, 0x0007), 0 },	/* Audigy LS or Live 24bit */
+	{ 0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
+
+// pci_driver definition
+static struct pci_driver ca0106_driver = {
+	.name = KBUILD_MODNAME,
+	.id_table = snd_ca0106_ids,
+	.probe = snd_ca0106_probe,
+	.driver = {
+		.pm = SND_CA0106_PM_OPS,
+	},
+};
+
+module_pci_driver(ca0106_driver);
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
new file mode 100644
index 0000000000..1d5a899b2c
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -0,0 +1,902 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/*
+ *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ *  Version: 0.0.18
+ *
+ *  FEATURES currently supported:
+ *    See ca0106_main.c for features.
+ * 
+ *  Changelog:
+ *    Support interrupts per period.
+ *    Removed noise from Center/LFE channel when in Analog mode.
+ *    Rename and remove mixer controls.
+ *  0.0.6
+ *    Use separate card based DMA buffer for periods table list.
+ *  0.0.7
+ *    Change remove and rename ctrls into lists.
+ *  0.0.8
+ *    Try to fix capture sources.
+ *  0.0.9
+ *    Fix AC3 output.
+ *    Enable S32_LE format support.
+ *  0.0.10
+ *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ *  0.0.11
+ *    Add Model name recognition.
+ *  0.0.12
+ *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ *    Remove redundent "voice" handling.
+ *  0.0.13
+ *    Single trigger call for multi channels.
+ *  0.0.14
+ *    Set limits based on what the sound card hardware can do.
+ *    playback periods_min=2, periods_max=8
+ *    capture hw constraints require period_size = n * 64 bytes.
+ *    playback hw constraints require period_size = n * 64 bytes.
+ *  0.0.15
+ *    Separated ca0106.c into separate functional .c files.
+ *  0.0.16
+ *    Modified Copyright message.
+ *  0.0.17
+ *    Implement Mic and Line in Capture.
+ *  0.0.18
+ *    Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
+ *
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
+ *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/tlv.h>
+#include <linux/io.h>
+
+#include "ca0106.h"
+
+static void ca0106_spdif_enable(struct snd_ca0106 *emu)
+{
+	unsigned int val;
+
+	if (emu->spdif_enable) {
+		/* Digital */
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
+		val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000;
+		snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+		val = inl(emu->port + CA0106_GPIO) & ~0x101;
+		outl(val, emu->port + CA0106_GPIO);
+
+	} else {
+		/* Analog */
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+		snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
+		val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000;
+		snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val);
+		val = inl(emu->port + CA0106_GPIO) | 0x101;
+		outl(val, emu->port + CA0106_GPIO);
+	}
+}
+
+static void ca0106_set_capture_source(struct snd_ca0106 *emu)
+{
+	unsigned int val = emu->capture_source;
+	unsigned int source, mask;
+	source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+	mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
+	snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+}
+
+static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu,
+					  unsigned int val, int force)
+{
+	unsigned int ngain, ogain;
+	u32 source;
+
+	snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+	ngain = emu->i2c_capture_volume[val][0]; /* Left */
+	ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+	if (force || ngain != ogain)
+		snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff);
+	ngain = emu->i2c_capture_volume[val][1]; /* Right */
+	ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */
+	if (force || ngain != ogain)
+		snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff);
+	source = 1 << val;
+	snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
+	emu->i2c_capture_source = val;
+}
+
+static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu)
+{
+	u32 tmp;
+
+	if (emu->capture_mic_line_in) {
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+		tmp = inl(emu->port + CA0106_GPIO) & ~0x400;
+		tmp = tmp | 0x400;
+		outl(tmp, emu->port + CA0106_GPIO);
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */
+	} else {
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */
+		tmp = inl(emu->port + CA0106_GPIO) & ~0x400;
+		outl(tmp, emu->port + CA0106_GPIO);
+		/* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */
+	}
+}
+
+static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx)
+{
+	snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]);
+}
+
+/*
+ */
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1);
+static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1);
+
+#define snd_ca0106_shared_spdif_info	snd_ctl_boolean_mono_info
+
+static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] = emu->spdif_enable;
+	return 0;
+}
+
+static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int val;
+	int change = 0;
+
+	val = !!ucontrol->value.integer.value[0];
+	change = (emu->spdif_enable != val);
+	if (change) {
+		emu->spdif_enable = val;
+		ca0106_spdif_enable(emu);
+	}
+        return change;
+}
+
+static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[6] = {
+		"IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, 6, texts);
+}
+
+static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = emu->capture_source;
+	return 0;
+}
+
+static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int val;
+	int change = 0;
+
+	val = ucontrol->value.enumerated.item[0] ;
+	if (val >= 6)
+		return -EINVAL;
+	change = (emu->capture_source != val);
+	if (change) {
+		emu->capture_source = val;
+		ca0106_set_capture_source(emu);
+	}
+        return change;
+}
+
+static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[4] = {
+		"Phone", "Mic", "Line in", "Aux"
+	};
+
+	return snd_ctl_enum_info(uinfo, 1, 4, texts);
+}
+
+static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = emu->i2c_capture_source;
+	return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int source_id;
+	int change = 0;
+	/* If the capture source has changed,
+	 * update the capture volume from the cached value
+	 * for the particular source.
+	 */
+	source_id = ucontrol->value.enumerated.item[0] ;
+	if (source_id >= 4)
+		return -EINVAL;
+	change = (emu->i2c_capture_source != source_id);
+	if (change) {
+		ca0106_set_i2c_capture_source(emu, source_id, 0);
+	}
+        return change;
+}
+
+static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol,
+					       struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[2] = { "Side out", "Line in" };
+
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol,
+					       struct snd_ctl_elem_info *uinfo)
+{
+	static const char * const texts[2] = { "Line in", "Mic in" };
+
+	return snd_ctl_enum_info(uinfo, 1, 2, texts);
+}
+
+static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = emu->capture_mic_line_in;
+	return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int val;
+	int change = 0;
+
+	val = ucontrol->value.enumerated.item[0] ;
+	if (val > 1)
+		return -EINVAL;
+	change = (emu->capture_mic_line_in != val);
+	if (change) {
+		emu->capture_mic_line_in = val;
+		ca0106_set_capture_mic_line_in(emu);
+	}
+        return change;
+}
+
+static const struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
+{
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name =		"Shared Mic/Line in Capture Switch",
+	.info =		snd_ca0106_capture_mic_line_in_info,
+	.get =		snd_ca0106_capture_mic_line_in_get,
+	.put =		snd_ca0106_capture_mic_line_in_put
+};
+
+static const struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out =
+{
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name =		"Shared Line in/Side out Capture Switch",
+	.info =		snd_ca0106_capture_line_in_side_out_info,
+	.get =		snd_ca0106_capture_mic_line_in_get,
+	.put =		snd_ca0106_capture_mic_line_in_put
+};
+
+
+static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+	return 0;
+}
+
+static void decode_spdif_bits(unsigned char *status, unsigned int bits)
+{
+	status[0] = (bits >> 0) & 0xff;
+	status[1] = (bits >> 8) & 0xff;
+	status[2] = (bits >> 16) & 0xff;
+	status[3] = (bits >> 24) & 0xff;
+}
+
+static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	decode_spdif_bits(ucontrol->value.iec958.status,
+			  emu->spdif_bits[idx]);
+        return 0;
+}
+
+static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	decode_spdif_bits(ucontrol->value.iec958.status,
+			  emu->spdif_str_bits[idx]);
+        return 0;
+}
+
+static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.iec958.status[0] = 0xff;
+	ucontrol->value.iec958.status[1] = 0xff;
+	ucontrol->value.iec958.status[2] = 0xff;
+	ucontrol->value.iec958.status[3] = 0xff;
+        return 0;
+}
+
+static unsigned int encode_spdif_bits(unsigned char *status)
+{
+	return ((unsigned int)status[0] << 0) |
+		((unsigned int)status[1] << 8) |
+		((unsigned int)status[2] << 16) |
+		((unsigned int)status[3] << 24);
+}
+
+static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	unsigned int val;
+
+	val = encode_spdif_bits(ucontrol->value.iec958.status);
+	if (val != emu->spdif_bits[idx]) {
+		emu->spdif_bits[idx] = val;
+		/* FIXME: this isn't safe, but needed to keep the compatibility
+		 * with older alsa-lib config
+		 */
+		emu->spdif_str_bits[idx] = val;
+		ca0106_set_spdif_bits(emu, idx);
+		return 1;
+	}
+	return 0;
+}
+
+static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	unsigned int val;
+
+	val = encode_spdif_bits(ucontrol->value.iec958.status);
+	if (val != emu->spdif_str_bits[idx]) {
+		emu->spdif_str_bits[idx] = val;
+		ca0106_set_spdif_bits(emu, idx);
+		return 1;
+	}
+        return 0;
+}
+
+static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 2;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 255;
+        return 0;
+}
+
+static int snd_ca0106_volume_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+        unsigned int value;
+	int channel_id, reg;
+
+	channel_id = (kcontrol->private_value >> 8) & 0xff;
+	reg = kcontrol->private_value & 0xff;
+
+        value = snd_ca0106_ptr_read(emu, reg, channel_id);
+        ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
+        ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
+        return 0;
+}
+
+static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+        unsigned int oval, nval;
+	int channel_id, reg;
+
+	channel_id = (kcontrol->private_value >> 8) & 0xff;
+	reg = kcontrol->private_value & 0xff;
+
+	oval = snd_ca0106_ptr_read(emu, reg, channel_id);
+	nval = ((0xff - ucontrol->value.integer.value[0]) << 24) |
+		((0xff - ucontrol->value.integer.value[1]) << 16);
+        nval |= ((0xff - ucontrol->value.integer.value[0]) << 8) |
+		((0xff - ucontrol->value.integer.value[1]) );
+	if (oval == nval)
+		return 0;
+	snd_ca0106_ptr_write(emu, reg, channel_id, nval);
+	return 1;
+}
+
+static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 2;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 255;
+        return 0;
+}
+
+static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	int source_id;
+
+	source_id = kcontrol->private_value;
+
+        ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0];
+        ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1];
+        return 0;
+}
+
+static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+        unsigned int ogain;
+        unsigned int ngain;
+	int source_id;
+	int change = 0;
+
+	source_id = kcontrol->private_value;
+	ogain = emu->i2c_capture_volume[source_id][0]; /* Left */
+	ngain = ucontrol->value.integer.value[0];
+	if (ngain > 0xff)
+		return -EINVAL;
+	if (ogain != ngain) {
+		if (emu->i2c_capture_source == source_id)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) );
+		emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0];
+		change = 1;
+	}
+	ogain = emu->i2c_capture_volume[source_id][1]; /* Right */
+	ngain = ucontrol->value.integer.value[1];
+	if (ngain > 0xff)
+		return -EINVAL;
+	if (ogain != ngain) {
+		if (emu->i2c_capture_source == source_id)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+		emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1];
+		change = 1;
+	}
+
+	return change;
+}
+
+#define spi_mute_info	snd_ctl_boolean_mono_info
+
+static int spi_mute_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+	unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+
+	ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit);
+	return 0;
+}
+
+static int spi_mute_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT;
+	unsigned int bit = kcontrol->private_value & SPI_REG_MASK;
+	int ret;
+
+	ret = emu->spi_dac_reg[reg] & bit;
+	if (ucontrol->value.integer.value[0]) {
+		if (!ret)	/* bit already cleared, do nothing */
+			return 0;
+		emu->spi_dac_reg[reg] &= ~bit;
+	} else {
+		if (ret)	/* bit already set, do nothing */
+			return 0;
+		emu->spi_dac_reg[reg] |= bit;
+	}
+
+	ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]);
+	return ret ? -EINVAL : 1;
+}
+
+#define CA_VOLUME(xname,chid,reg) \
+{								\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,	\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |		\
+	          SNDRV_CTL_ELEM_ACCESS_TLV_READ,		\
+	.info =	 snd_ca0106_volume_info,			\
+	.get =   snd_ca0106_volume_get,				\
+	.put =   snd_ca0106_volume_put,				\
+	.tlv = { .p = snd_ca0106_db_scale1 },			\
+	.private_value = ((chid) << 8) | (reg)			\
+}
+
+static const struct snd_kcontrol_new snd_ca0106_volume_ctls[] = {
+	CA_VOLUME("Analog Front Playback Volume",
+		  CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2),
+        CA_VOLUME("Analog Rear Playback Volume",
+		  CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2),
+	CA_VOLUME("Analog Center/LFE Playback Volume",
+		  CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2),
+        CA_VOLUME("Analog Side Playback Volume",
+		  CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2),
+
+        CA_VOLUME("IEC958 Front Playback Volume",
+		  CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1),
+	CA_VOLUME("IEC958 Rear Playback Volume",
+		  CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1),
+	CA_VOLUME("IEC958 Center/LFE Playback Volume",
+		  CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1),
+	CA_VOLUME("IEC958 Unknown Playback Volume",
+		  CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1),
+
+        CA_VOLUME("CAPTURE feedback Playback Volume",
+		  1, CAPTURE_CONTROL),
+
+	{
+		.access =	SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
+		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+		.count =	4,
+		.info =         snd_ca0106_spdif_info,
+		.get =          snd_ca0106_spdif_get_mask
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name =		"IEC958 Playback Switch",
+		.info =		snd_ca0106_shared_spdif_info,
+		.get =		snd_ca0106_shared_spdif_get,
+		.put =		snd_ca0106_shared_spdif_put
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name =		"Digital Source Capture Enum",
+		.info =		snd_ca0106_capture_source_info,
+		.get =		snd_ca0106_capture_source_get,
+		.put =		snd_ca0106_capture_source_put
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name =		"Analog Source Capture Enum",
+		.info =		snd_ca0106_i2c_capture_source_info,
+		.get =		snd_ca0106_i2c_capture_source_get,
+		.put =		snd_ca0106_i2c_capture_source_put
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
+		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+		.count =	4,
+		.info =         snd_ca0106_spdif_info,
+		.get =          snd_ca0106_spdif_get_default,
+		.put =          snd_ca0106_spdif_put_default
+	},
+	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
+		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM),
+		.count =	4,
+		.info =         snd_ca0106_spdif_info,
+		.get =          snd_ca0106_spdif_get_stream,
+		.put =          snd_ca0106_spdif_put_stream
+	},
+};
+
+#define I2C_VOLUME(xname,chid) \
+{								\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,	\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |		\
+	          SNDRV_CTL_ELEM_ACCESS_TLV_READ,		\
+	.info =  snd_ca0106_i2c_volume_info,			\
+	.get =   snd_ca0106_i2c_volume_get,			\
+	.put =   snd_ca0106_i2c_volume_put,			\
+	.tlv = { .p = snd_ca0106_db_scale2 },			\
+	.private_value = chid					\
+}
+
+static const struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] = {
+        I2C_VOLUME("Phone Capture Volume", 0),
+        I2C_VOLUME("Mic Capture Volume", 1),
+        I2C_VOLUME("Line in Capture Volume", 2),
+        I2C_VOLUME("Aux Capture Volume", 3),
+};
+
+static const int spi_dmute_reg[] = {
+	SPI_DMUTE0_REG,
+	SPI_DMUTE1_REG,
+	SPI_DMUTE2_REG,
+	0,
+	SPI_DMUTE4_REG,
+};
+static const int spi_dmute_bit[] = {
+	SPI_DMUTE0_BIT,
+	SPI_DMUTE1_BIT,
+	SPI_DMUTE2_BIT,
+	0,
+	SPI_DMUTE4_BIT,
+};
+
+static struct snd_kcontrol_new
+snd_ca0106_volume_spi_dac_ctl(const struct snd_ca0106_details *details,
+			      int channel_id)
+{
+	struct snd_kcontrol_new spi_switch = {0};
+	int reg, bit;
+	int dac_id;
+
+	spi_switch.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	spi_switch.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+	spi_switch.info = spi_mute_info;
+	spi_switch.get = spi_mute_get;
+	spi_switch.put = spi_mute_put;
+
+	switch (channel_id) {
+	case PCM_FRONT_CHANNEL:
+		spi_switch.name = "Analog Front Playback Switch";
+		dac_id = (details->spi_dac & 0xf000) >> (4 * 3);
+		break;
+	case PCM_REAR_CHANNEL:
+		spi_switch.name = "Analog Rear Playback Switch";
+		dac_id = (details->spi_dac & 0x0f00) >> (4 * 2);
+		break;
+	case PCM_CENTER_LFE_CHANNEL:
+		spi_switch.name = "Analog Center/LFE Playback Switch";
+		dac_id = (details->spi_dac & 0x00f0) >> (4 * 1);
+		break;
+	case PCM_UNKNOWN_CHANNEL:
+		spi_switch.name = "Analog Side Playback Switch";
+		dac_id = (details->spi_dac & 0x000f) >> (4 * 0);
+		break;
+	default:
+		/* Unused channel */
+		spi_switch.name = NULL;
+		dac_id = 0;
+	}
+	reg = spi_dmute_reg[dac_id];
+	bit = spi_dmute_bit[dac_id];
+
+	spi_switch.private_value = (reg << SPI_REG_SHIFT) | bit;
+
+	return spi_switch;
+}
+
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	strcpy(id.name, name);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_remove_id(card, &id);
+}
+
+static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
+{
+	struct snd_kcontrol *kctl = snd_ctl_find_id_mixer(card, src);
+	if (kctl) {
+		snd_ctl_rename(card, kctl, dst);
+		return 0;
+	}
+	return -ENOENT;
+}
+
+#define ADD_CTLS(emu, ctls)						\
+	do {								\
+		int i, _err;						\
+		for (i = 0; i < ARRAY_SIZE(ctls); i++) {		\
+			_err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \
+			if (_err < 0)					\
+				return _err;				\
+		}							\
+	} while (0)
+
+static
+DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
+
+static const char * const follower_vols[] = {
+	"Analog Front Playback Volume",
+        "Analog Rear Playback Volume",
+	"Analog Center/LFE Playback Volume",
+        "Analog Side Playback Volume",
+        "IEC958 Front Playback Volume",
+	"IEC958 Rear Playback Volume",
+	"IEC958 Center/LFE Playback Volume",
+	"IEC958 Unknown Playback Volume",
+        "CAPTURE feedback Playback Volume",
+	NULL
+};
+
+static const char * const follower_sws[] = {
+	"Analog Front Playback Switch",
+	"Analog Rear Playback Switch",
+	"Analog Center/LFE Playback Switch",
+	"Analog Side Playback Switch",
+	"IEC958 Playback Switch",
+	NULL
+};
+
+int snd_ca0106_mixer(struct snd_ca0106 *emu)
+{
+	int err;
+        struct snd_card *card = emu->card;
+	const char * const *c;
+	struct snd_kcontrol *vmaster;
+	static const char * const ca0106_remove_ctls[] = {
+		"Master Mono Playback Switch",
+		"Master Mono Playback Volume",
+		"3D Control - Switch",
+		"3D Control Sigmatel - Depth",
+		"PCM Playback Switch",
+		"PCM Playback Volume",
+		"CD Playback Switch",
+		"CD Playback Volume",
+		"Phone Playback Switch",
+		"Phone Playback Volume",
+		"Video Playback Switch",
+		"Video Playback Volume",
+		"Beep Playback Switch",
+		"Beep Playback Volume",
+		"Mono Output Select",
+		"Capture Source",
+		"Capture Switch",
+		"Capture Volume",
+		"External Amplifier",
+		"Sigmatel 4-Speaker Stereo Playback Switch",
+		"Surround Phase Inversion Playback Switch",
+		NULL
+	};
+	static const char * const ca0106_rename_ctls[] = {
+		"Master Playback Switch", "Capture Switch",
+		"Master Playback Volume", "Capture Volume",
+		"Line Playback Switch", "AC97 Line Capture Switch",
+		"Line Playback Volume", "AC97 Line Capture Volume",
+		"Aux Playback Switch", "AC97 Aux Capture Switch",
+		"Aux Playback Volume", "AC97 Aux Capture Volume",
+		"Mic Playback Switch", "AC97 Mic Capture Switch",
+		"Mic Playback Volume", "AC97 Mic Capture Volume",
+		"Mic Select", "AC97 Mic Select",
+		"Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)",
+		NULL
+	};
+#if 1
+	for (c = ca0106_remove_ctls; *c; c++)
+		remove_ctl(card, *c);
+	for (c = ca0106_rename_ctls; *c; c += 2)
+		rename_ctl(card, c[0], c[1]);
+#endif
+
+	ADD_CTLS(emu, snd_ca0106_volume_ctls);
+	if (emu->details->i2c_adc == 1) {
+		ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls);
+		if (emu->details->gpio_type == 1)
+			err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
+		else  /* gpio_type == 2 */
+			err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu));
+		if (err < 0)
+			return err;
+	}
+	if (emu->details->spi_dac) {
+		int i;
+		for (i = 0;; i++) {
+			struct snd_kcontrol_new ctl;
+			ctl = snd_ca0106_volume_spi_dac_ctl(emu->details, i);
+			if (!ctl.name)
+				break;
+			err = snd_ctl_add(card, snd_ctl_new1(&ctl, emu));
+			if (err < 0)
+				return err;
+		}
+	}
+
+	/* Create virtual master controls */
+	vmaster = snd_ctl_make_virtual_master("Master Playback Volume",
+					      snd_ca0106_master_db_scale);
+	if (!vmaster)
+		return -ENOMEM;
+	err = snd_ctl_add(card, vmaster);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_followers(card, vmaster, follower_vols);
+	if (err < 0)
+		return err;
+
+	if (emu->details->spi_dac) {
+		vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
+						      NULL);
+		if (!vmaster)
+			return -ENOMEM;
+		err = snd_ctl_add(card, vmaster);
+		if (err < 0)
+			return err;
+		err = snd_ctl_add_followers(card, vmaster, follower_sws);
+		if (err < 0)
+			return err;
+	}
+
+	strcpy(card->mixername, "CA0106");
+        return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+struct ca0106_vol_tbl {
+	unsigned int channel_id;
+	unsigned int reg;
+};
+
+static const struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = {
+	{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 },
+	{ CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 },
+	{ CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 },
+	{ 1, CAPTURE_CONTROL },
+};
+
+void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip)
+{
+	int i;
+
+	/* save volumes */
+	for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+		chip->saved_vol[i] =
+			snd_ca0106_ptr_read(chip, saved_volumes[i].reg,
+					    saved_volumes[i].channel_id);
+}
+
+void snd_ca0106_mixer_resume(struct snd_ca0106  *chip)
+{
+	int i;
+
+	for (i = 0; i < NUM_SAVED_VOLUMES; i++)
+		snd_ca0106_ptr_write(chip, saved_volumes[i].reg,
+				     saved_volumes[i].channel_id,
+				     chip->saved_vol[i]);
+
+	ca0106_spdif_enable(chip);
+	ca0106_set_capture_source(chip);
+	ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1);
+	for (i = 0; i < 4; i++)
+		ca0106_set_spdif_bits(chip, i);
+	if (chip->details->i2c_adc)
+		ca0106_set_capture_mic_line_in(chip);
+}
+#endif /* CONFIG_PM_SLEEP */
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
new file mode 100644
index 0000000000..c99603e137
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -0,0 +1,429 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/*
+ *  Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ *  Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ *  Version: 0.0.18
+ *
+ *  FEATURES currently supported:
+ *    See ca0106_main.c for features.
+ * 
+ *  Changelog:
+ *    Support interrupts per period.
+ *    Removed noise from Center/LFE channel when in Analog mode.
+ *    Rename and remove mixer controls.
+ *  0.0.6
+ *    Use separate card based DMA buffer for periods table list.
+ *  0.0.7
+ *    Change remove and rename ctrls into lists.
+ *  0.0.8
+ *    Try to fix capture sources.
+ *  0.0.9
+ *    Fix AC3 output.
+ *    Enable S32_LE format support.
+ *  0.0.10
+ *    Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
+ *  0.0.11
+ *    Add Model name recognition.
+ *  0.0.12
+ *    Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
+ *    Remove redundent "voice" handling.
+ *  0.0.13
+ *    Single trigger call for multi channels.
+ *  0.0.14
+ *    Set limits based on what the sound card hardware can do.
+ *    playback periods_min=2, periods_max=8
+ *    capture hw constraints require period_size = n * 64 bytes.
+ *    playback hw constraints require period_size = n * 64 bytes.
+ *  0.0.15
+ *    Separate ca0106.c into separate functional .c files.
+ *  0.0.16
+ *    Modified Copyright message.
+ *  0.0.17
+ *    Add iec958 file in proc file system to show status of SPDIF in.
+ *  0.0.18
+ *    Implement support for Line-in capture on SB Live 24bit.
+ *
+ *  This code was initially based on code from ALSA's emu10k1x.c which is:
+ *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ */
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+
+#include "ca0106.h"
+
+
+struct snd_ca0106_category_str {
+	int val;
+	const char *name;
+};
+
+static const struct snd_ca0106_category_str snd_ca0106_con_category[] = {
+	{ IEC958_AES1_CON_DAT, "DAT" },
+	{ IEC958_AES1_CON_VCR, "VCR" },
+	{ IEC958_AES1_CON_MICROPHONE, "microphone" },
+	{ IEC958_AES1_CON_SYNTHESIZER, "synthesizer" },
+	{ IEC958_AES1_CON_RATE_CONVERTER, "rate converter" },
+	{ IEC958_AES1_CON_MIXER, "mixer" },
+	{ IEC958_AES1_CON_SAMPLER, "sampler" },
+	{ IEC958_AES1_CON_PCM_CODER, "PCM coder" },
+	{ IEC958_AES1_CON_IEC908_CD, "CD" },
+	{ IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" },
+	{ IEC958_AES1_CON_GENERAL, "general" },
+};
+
+
+static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 value)
+{
+	int i;
+	u32 status[4];
+	status[0] = value & 0xff;
+	status[1] = (value >> 8) & 0xff;
+	status[2] = (value >> 16)  & 0xff;
+	status[3] = (value >> 24)  & 0xff;
+	
+	if (! (status[0] & IEC958_AES0_PROFESSIONAL)) {
+		/* consumer */
+		snd_iprintf(buffer, "Mode: consumer\n");
+		snd_iprintf(buffer, "Data: ");
+		if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+			snd_iprintf(buffer, "audio\n");
+		} else {
+			snd_iprintf(buffer, "non-audio\n");
+		}
+		snd_iprintf(buffer, "Rate: ");
+		switch (status[3] & IEC958_AES3_CON_FS) {
+		case IEC958_AES3_CON_FS_44100:
+			snd_iprintf(buffer, "44100 Hz\n");
+			break;
+		case IEC958_AES3_CON_FS_48000:
+			snd_iprintf(buffer, "48000 Hz\n");
+			break;
+		case IEC958_AES3_CON_FS_32000:
+			snd_iprintf(buffer, "32000 Hz\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+		snd_iprintf(buffer, "Copyright: ");
+		if (status[0] & IEC958_AES0_CON_NOT_COPYRIGHT) {
+			snd_iprintf(buffer, "permitted\n");
+		} else {
+			snd_iprintf(buffer, "protected\n");
+		}
+		snd_iprintf(buffer, "Emphasis: ");
+		if ((status[0] & IEC958_AES0_CON_EMPHASIS) != IEC958_AES0_CON_EMPHASIS_5015) {
+			snd_iprintf(buffer, "none\n");
+		} else {
+			snd_iprintf(buffer, "50/15us\n");
+		}
+		snd_iprintf(buffer, "Category: ");
+		for (i = 0; i < ARRAY_SIZE(snd_ca0106_con_category); i++) {
+			if ((status[1] & IEC958_AES1_CON_CATEGORY) == snd_ca0106_con_category[i].val) {
+				snd_iprintf(buffer, "%s\n", snd_ca0106_con_category[i].name);
+				break;
+			}
+		}
+		if (i >= ARRAY_SIZE(snd_ca0106_con_category)) {
+			snd_iprintf(buffer, "unknown 0x%x\n", status[1] & IEC958_AES1_CON_CATEGORY);
+		}
+		snd_iprintf(buffer, "Original: ");
+		if (status[1] & IEC958_AES1_CON_ORIGINAL) {
+			snd_iprintf(buffer, "original\n");
+		} else {
+			snd_iprintf(buffer, "1st generation\n");
+		}
+		snd_iprintf(buffer, "Clock: ");
+		switch (status[3] & IEC958_AES3_CON_CLOCK) {
+		case IEC958_AES3_CON_CLOCK_1000PPM:
+			snd_iprintf(buffer, "1000 ppm\n");
+			break;
+		case IEC958_AES3_CON_CLOCK_50PPM:
+			snd_iprintf(buffer, "50 ppm\n");
+			break;
+		case IEC958_AES3_CON_CLOCK_VARIABLE:
+			snd_iprintf(buffer, "variable pitch\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+	} else {
+		snd_iprintf(buffer, "Mode: professional\n");
+		snd_iprintf(buffer, "Data: ");
+		if (!(status[0] & IEC958_AES0_NONAUDIO)) {
+			snd_iprintf(buffer, "audio\n");
+		} else {
+			snd_iprintf(buffer, "non-audio\n");
+		}
+		snd_iprintf(buffer, "Rate: ");
+		switch (status[0] & IEC958_AES0_PRO_FS) {
+		case IEC958_AES0_PRO_FS_44100:
+			snd_iprintf(buffer, "44100 Hz\n");
+			break;
+		case IEC958_AES0_PRO_FS_48000:
+			snd_iprintf(buffer, "48000 Hz\n");
+			break;
+		case IEC958_AES0_PRO_FS_32000:
+			snd_iprintf(buffer, "32000 Hz\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+		snd_iprintf(buffer, "Rate Locked: ");
+		if (status[0] & IEC958_AES0_PRO_FREQ_UNLOCKED)
+			snd_iprintf(buffer, "no\n");
+		else
+			snd_iprintf(buffer, "yes\n");
+		snd_iprintf(buffer, "Emphasis: ");
+		switch (status[0] & IEC958_AES0_PRO_EMPHASIS) {
+		case IEC958_AES0_PRO_EMPHASIS_CCITT:
+			snd_iprintf(buffer, "CCITT J.17\n");
+			break;
+		case IEC958_AES0_PRO_EMPHASIS_NONE:
+			snd_iprintf(buffer, "none\n");
+			break;
+		case IEC958_AES0_PRO_EMPHASIS_5015:
+			snd_iprintf(buffer, "50/15us\n");
+			break;
+		case IEC958_AES0_PRO_EMPHASIS_NOTID:
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+		snd_iprintf(buffer, "Stereophonic: ");
+		if ((status[1] & IEC958_AES1_PRO_MODE) == IEC958_AES1_PRO_MODE_STEREOPHONIC) {
+			snd_iprintf(buffer, "stereo\n");
+		} else {
+			snd_iprintf(buffer, "not indicated\n");
+		}
+		snd_iprintf(buffer, "Userbits: ");
+		switch (status[1] & IEC958_AES1_PRO_USERBITS) {
+		case IEC958_AES1_PRO_USERBITS_192:
+			snd_iprintf(buffer, "192bit\n");
+			break;
+		case IEC958_AES1_PRO_USERBITS_UDEF:
+			snd_iprintf(buffer, "user-defined\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+		snd_iprintf(buffer, "Sample Bits: ");
+		switch (status[2] & IEC958_AES2_PRO_SBITS) {
+		case IEC958_AES2_PRO_SBITS_20:
+			snd_iprintf(buffer, "20 bit\n");
+			break;
+		case IEC958_AES2_PRO_SBITS_24:
+			snd_iprintf(buffer, "24 bit\n");
+			break;
+		case IEC958_AES2_PRO_SBITS_UDEF:
+			snd_iprintf(buffer, "user defined\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+		snd_iprintf(buffer, "Word Length: ");
+		switch (status[2] & IEC958_AES2_PRO_WORDLEN) {
+		case IEC958_AES2_PRO_WORDLEN_22_18:
+			snd_iprintf(buffer, "22 bit or 18 bit\n");
+			break;
+		case IEC958_AES2_PRO_WORDLEN_23_19:
+			snd_iprintf(buffer, "23 bit or 19 bit\n");
+			break;
+		case IEC958_AES2_PRO_WORDLEN_24_20:
+			snd_iprintf(buffer, "24 bit or 20 bit\n");
+			break;
+		case IEC958_AES2_PRO_WORDLEN_20_16:
+			snd_iprintf(buffer, "20 bit or 16 bit\n");
+			break;
+		default:
+			snd_iprintf(buffer, "unknown\n");
+			break;
+		}
+	}
+}
+
+static void snd_ca0106_proc_iec958(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	u32 value;
+
+        value = snd_ca0106_ptr_read(emu, SAMPLE_RATE_TRACKER_STATUS, 0);
+	snd_iprintf(buffer, "Status: %s, %s, %s\n",
+		  (value & 0x100000) ? "Rate Locked" : "Not Rate Locked",
+		  (value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock",
+		  (value & 0x400000) ? "Audio Valid" : "No valid audio" );
+	snd_iprintf(buffer, "Estimated sample rate: %u\n", 
+		  ((value & 0xfffff) * 48000) / 0x8000 );
+	if (value & 0x200000) {
+		snd_iprintf(buffer, "IEC958/SPDIF input status:\n");
+        	value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0);
+		snd_ca0106_proc_dump_iec958(buffer, value);
+	}
+
+	snd_iprintf(buffer, "\n");
+}
+
+static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	unsigned long flags;
+        char line[64];
+        u32 reg, val;
+        while (!snd_info_get_line(buffer, line, sizeof(line))) {
+                if (sscanf(line, "%x %x", &reg, &val) != 2)
+                        continue;
+		if (reg < 0x40 && val <= 0xffffffff) {
+			spin_lock_irqsave(&emu->emu_lock, flags);
+			outl(val, emu->port + (reg & 0xfffffffc));
+			spin_unlock_irqrestore(&emu->emu_lock, flags);
+		}
+        }
+}
+
+static void snd_ca0106_proc_reg_read32(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	unsigned long value;
+	unsigned long flags;
+	int i;
+	snd_iprintf(buffer, "Registers:\n\n");
+	for(i = 0; i < 0x20; i+=4) {
+		spin_lock_irqsave(&emu->emu_lock, flags);
+		value = inl(emu->port + i);
+		spin_unlock_irqrestore(&emu->emu_lock, flags);
+		snd_iprintf(buffer, "Register %02X: %08lX\n", i, value);
+	}
+}
+
+static void snd_ca0106_proc_reg_read16(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+        unsigned int value;
+	unsigned long flags;
+	int i;
+	snd_iprintf(buffer, "Registers:\n\n");
+	for(i = 0; i < 0x20; i+=2) {
+		spin_lock_irqsave(&emu->emu_lock, flags);
+		value = inw(emu->port + i);
+		spin_unlock_irqrestore(&emu->emu_lock, flags);
+		snd_iprintf(buffer, "Register %02X: %04X\n", i, value);
+	}
+}
+
+static void snd_ca0106_proc_reg_read8(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	unsigned int value;
+	unsigned long flags;
+	int i;
+	snd_iprintf(buffer, "Registers:\n\n");
+	for(i = 0; i < 0x20; i+=1) {
+		spin_lock_irqsave(&emu->emu_lock, flags);
+		value = inb(emu->port + i);
+		spin_unlock_irqrestore(&emu->emu_lock, flags);
+		snd_iprintf(buffer, "Register %02X: %02X\n", i, value);
+	}
+}
+
+static void snd_ca0106_proc_reg_read1(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	unsigned long value;
+	int i,j;
+
+	snd_iprintf(buffer, "Registers\n");
+	for(i = 0; i < 0x40; i++) {
+		snd_iprintf(buffer, "%02X: ",i);
+		for (j = 0; j < 4; j++) {
+                  value = snd_ca0106_ptr_read(emu, i, j);
+		  snd_iprintf(buffer, "%08lX ", value);
+                }
+	        snd_iprintf(buffer, "\n");
+	}
+}
+
+static void snd_ca0106_proc_reg_read2(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+	unsigned long value;
+	int i,j;
+
+	snd_iprintf(buffer, "Registers\n");
+	for(i = 0x40; i < 0x80; i++) {
+		snd_iprintf(buffer, "%02X: ",i);
+		for (j = 0; j < 4; j++) {
+                  value = snd_ca0106_ptr_read(emu, i, j);
+		  snd_iprintf(buffer, "%08lX ", value);
+                }
+	        snd_iprintf(buffer, "\n");
+	}
+}
+
+static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+        char line[64];
+        unsigned int reg, channel_id , val;
+        while (!snd_info_get_line(buffer, line, sizeof(line))) {
+                if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
+                        continue;
+		if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3)
+                        snd_ca0106_ptr_write(emu, reg, channel_id, val);
+        }
+}
+
+static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry, 
+				       struct snd_info_buffer *buffer)
+{
+	struct snd_ca0106 *emu = entry->private_data;
+        char line[64];
+        unsigned int reg, val;
+        while (!snd_info_get_line(buffer, line, sizeof(line))) {
+                if (sscanf(line, "%x %x", &reg, &val) != 2)
+                        continue;
+                if ((reg <= 0x7f) || (val <= 0x1ff)) {
+                        snd_ca0106_i2c_write(emu, reg, val);
+		}
+        }
+}
+
+int snd_ca0106_proc_init(struct snd_ca0106 *emu)
+{
+	snd_card_ro_proc_new(emu->card, "iec958", emu, snd_ca0106_proc_iec958);
+	snd_card_rw_proc_new(emu->card, "ca0106_reg32", emu,
+			     snd_ca0106_proc_reg_read32,
+			     snd_ca0106_proc_reg_write32);
+	snd_card_ro_proc_new(emu->card, "ca0106_reg16", emu,
+			     snd_ca0106_proc_reg_read16);
+	snd_card_ro_proc_new(emu->card, "ca0106_reg8", emu,
+			     snd_ca0106_proc_reg_read8);
+	snd_card_rw_proc_new(emu->card, "ca0106_regs1", emu,
+			     snd_ca0106_proc_reg_read1,
+			     snd_ca0106_proc_reg_write);
+	snd_card_rw_proc_new(emu->card, "ca0106_i2c", emu, NULL,
+			     snd_ca0106_proc_i2c_write);
+	snd_card_ro_proc_new(emu->card, "ca0106_regs2", emu,
+			     snd_ca0106_proc_reg_read2);
+	return 0;
+}
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
new file mode 100644
index 0000000000..957e60f648
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.c
@@ -0,0 +1,302 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/* 
+ *  Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ *  Creative Audio MIDI, for the CA0106 Driver
+ *  Version: 0.0.1
+ *
+ *  Changelog:
+ *    Implementation is based on mpu401 and emu10k1x and
+ *    tested with ca0106.
+ *    mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *    emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ */
+
+#include <linux/spinlock.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+
+#include "ca_midi.h"
+
+#define ca_midi_write_data(midi, data)	midi->write(midi, data, 0)
+#define ca_midi_write_cmd(midi, data)	midi->write(midi, data, 1)
+#define ca_midi_read_data(midi)		midi->read(midi, 0)
+#define ca_midi_read_stat(midi)		midi->read(midi, 1)
+#define ca_midi_input_avail(midi)	(!(ca_midi_read_stat(midi) & midi->input_avail))
+#define ca_midi_output_ready(midi)	(!(ca_midi_read_stat(midi) & midi->output_ready))
+
+static void ca_midi_clear_rx(struct snd_ca_midi *midi)
+{
+	int timeout = 100000;
+	for (; timeout > 0 && ca_midi_input_avail(midi); timeout--)
+		ca_midi_read_data(midi);
+#ifdef CONFIG_SND_DEBUG
+	if (timeout <= 0)
+		pr_err("ca_midi_clear_rx: timeout (status = 0x%x)\n",
+			   ca_midi_read_stat(midi));
+#endif
+}
+
+static void ca_midi_interrupt(struct snd_ca_midi *midi, unsigned int status)
+{
+	unsigned char byte;
+
+	if (midi->rmidi == NULL) {
+		midi->interrupt_disable(midi,midi->tx_enable | midi->rx_enable);
+		return;
+	}
+
+	spin_lock(&midi->input_lock);
+	if ((status & midi->ipr_rx) && ca_midi_input_avail(midi)) {
+		if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+			ca_midi_clear_rx(midi);
+		} else {
+			byte = ca_midi_read_data(midi);
+			if(midi->substream_input)
+				snd_rawmidi_receive(midi->substream_input, &byte, 1);
+
+
+		}
+	}
+	spin_unlock(&midi->input_lock);
+
+	spin_lock(&midi->output_lock);
+	if ((status & midi->ipr_tx) && ca_midi_output_ready(midi)) {
+		if (midi->substream_output &&
+		    snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) {
+			ca_midi_write_data(midi, byte);
+		} else {
+			midi->interrupt_disable(midi,midi->tx_enable);
+		}
+	}
+	spin_unlock(&midi->output_lock);
+
+}
+
+static void ca_midi_cmd(struct snd_ca_midi *midi, unsigned char cmd, int ack)
+{
+	unsigned long flags;
+	int timeout, ok;
+
+	spin_lock_irqsave(&midi->input_lock, flags);
+	ca_midi_write_data(midi, 0x00);
+	/* ca_midi_clear_rx(midi); */
+
+	ca_midi_write_cmd(midi, cmd);
+	if (ack) {
+		ok = 0;
+		timeout = 10000;
+		while (!ok && timeout-- > 0) {
+			if (ca_midi_input_avail(midi)) {
+				if (ca_midi_read_data(midi) == midi->ack)
+					ok = 1;
+			}
+		}
+		if (!ok && ca_midi_read_data(midi) == midi->ack)
+			ok = 1;
+	} else {
+		ok = 1;
+	}
+	spin_unlock_irqrestore(&midi->input_lock, flags);
+	if (!ok)
+		pr_err("ca_midi_cmd: 0x%x failed at 0x%x (status = 0x%x, data = 0x%x)!!!\n",
+			   cmd,
+			   midi->get_dev_id_port(midi->dev_id),
+			   ca_midi_read_stat(midi),
+			   ca_midi_read_data(midi));
+}
+
+static int ca_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+	unsigned long flags;
+	
+	if (snd_BUG_ON(!midi->dev_id))
+		return -ENXIO;
+	spin_lock_irqsave(&midi->open_lock, flags);
+	midi->midi_mode |= CA_MIDI_MODE_INPUT;
+	midi->substream_input = substream;
+	if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+		ca_midi_cmd(midi, midi->reset, 1);
+		ca_midi_cmd(midi, midi->enter_uart, 1);
+	} else {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+	}
+	return 0;
+}
+
+static int ca_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+	unsigned long flags;
+
+	if (snd_BUG_ON(!midi->dev_id))
+		return -ENXIO;
+	spin_lock_irqsave(&midi->open_lock, flags);
+	midi->midi_mode |= CA_MIDI_MODE_OUTPUT;
+	midi->substream_output = substream;
+	if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+		ca_midi_cmd(midi, midi->reset, 1);
+		ca_midi_cmd(midi, midi->enter_uart, 1);
+	} else {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+	}
+	return 0;
+}
+
+static int ca_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+	unsigned long flags;
+
+	if (snd_BUG_ON(!midi->dev_id))
+		return -ENXIO;
+	spin_lock_irqsave(&midi->open_lock, flags);
+	midi->interrupt_disable(midi,midi->rx_enable);
+	midi->midi_mode &= ~CA_MIDI_MODE_INPUT;
+	midi->substream_input = NULL;
+	if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+		ca_midi_cmd(midi, midi->reset, 0);
+	} else {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+	}
+	return 0;
+}
+
+static int ca_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+	unsigned long flags;
+
+	if (snd_BUG_ON(!midi->dev_id))
+		return -ENXIO;
+	
+	spin_lock_irqsave(&midi->open_lock, flags);
+
+	midi->interrupt_disable(midi,midi->tx_enable);
+	midi->midi_mode &= ~CA_MIDI_MODE_OUTPUT;
+	midi->substream_output = NULL;
+	
+	if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+		ca_midi_cmd(midi, midi->reset, 0);
+	} else {
+		spin_unlock_irqrestore(&midi->open_lock, flags);
+	}
+	return 0;
+}
+
+static void ca_midi_input_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+
+	if (snd_BUG_ON(!midi->dev_id))
+		return;
+
+	if (up) {
+		midi->interrupt_enable(midi,midi->rx_enable);
+	} else {
+		midi->interrupt_disable(midi, midi->rx_enable);
+	}
+}
+
+static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
+{
+	struct snd_ca_midi *midi = substream->rmidi->private_data;
+	unsigned long flags;
+
+	if (snd_BUG_ON(!midi->dev_id))
+		return;
+
+	if (up) {
+		int max = 4;
+		unsigned char byte;
+
+		spin_lock_irqsave(&midi->output_lock, flags);
+	
+		/* try to send some amount of bytes here before interrupts */
+		while (max > 0) {
+			if (ca_midi_output_ready(midi)) {
+				if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT) ||
+				    snd_rawmidi_transmit(substream, &byte, 1) != 1) {
+					/* no more data */
+					spin_unlock_irqrestore(&midi->output_lock, flags);
+					return;
+				}
+				ca_midi_write_data(midi, byte);
+				max--;
+			} else {
+				break;
+			}
+		}
+
+		spin_unlock_irqrestore(&midi->output_lock, flags);
+		midi->interrupt_enable(midi,midi->tx_enable);
+
+	} else {
+		midi->interrupt_disable(midi,midi->tx_enable);
+	}
+}
+
+static const struct snd_rawmidi_ops ca_midi_output =
+{
+	.open =		ca_midi_output_open,
+	.close =	ca_midi_output_close,
+	.trigger =	ca_midi_output_trigger,
+};
+
+static const struct snd_rawmidi_ops ca_midi_input =
+{
+	.open =		ca_midi_input_open,
+	.close =	ca_midi_input_close,
+	.trigger =	ca_midi_input_trigger,
+};
+
+static void ca_midi_free(struct snd_ca_midi *midi)
+{
+	midi->interrupt = NULL;
+	midi->interrupt_enable = NULL;
+	midi->interrupt_disable = NULL;
+	midi->read = NULL;
+	midi->write = NULL;
+	midi->get_dev_id_card = NULL;
+	midi->get_dev_id_port = NULL;
+	midi->rmidi = NULL;
+}
+
+static void ca_rmidi_free(struct snd_rawmidi *rmidi)
+{
+	ca_midi_free(rmidi->private_data);
+}
+
+int ca_midi_init(void *dev_id, struct snd_ca_midi *midi, int device, char *name)
+{
+	struct snd_rawmidi *rmidi;
+	int err;
+
+	err = snd_rawmidi_new(midi->get_dev_id_card(midi->dev_id), name, device, 1, 1, &rmidi);
+	if (err < 0)
+		return err;
+
+	midi->dev_id = dev_id;
+	midi->interrupt = ca_midi_interrupt;
+
+	spin_lock_init(&midi->open_lock);
+	spin_lock_init(&midi->input_lock);
+	spin_lock_init(&midi->output_lock);
+
+	strcpy(rmidi->name, name);
+	snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &ca_midi_output);
+	snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &ca_midi_input);
+	rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+	                     SNDRV_RAWMIDI_INFO_INPUT |
+	                     SNDRV_RAWMIDI_INFO_DUPLEX;
+	rmidi->private_data = midi;
+	rmidi->private_free = ca_rmidi_free;
+	
+	midi->rmidi = rmidi;
+	return 0;
+}
+
diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h
new file mode 100644
index 0000000000..1d0476c8af
--- /dev/null
+++ b/sound/pci/ca0106/ca_midi.h
@@ -0,0 +1,52 @@
+/* SPDX-License-Identifier: GPL-2.0-or-later */
+/* 
+ *  Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de>
+ *  Creative Audio MIDI, for the CA0106 Driver
+ *  Version: 0.0.1
+ *
+ *  Changelog:
+ *    See ca_midi.c
+ */
+
+#include <linux/spinlock.h>
+#include <sound/rawmidi.h>
+#include <sound/mpu401.h>
+
+#define CA_MIDI_MODE_INPUT	MPU401_MODE_INPUT
+#define CA_MIDI_MODE_OUTPUT	MPU401_MODE_OUTPUT
+
+struct snd_ca_midi {
+
+	struct snd_rawmidi *rmidi;
+	struct snd_rawmidi_substream *substream_input;
+	struct snd_rawmidi_substream *substream_output;
+
+	void *dev_id;
+
+	spinlock_t input_lock;
+	spinlock_t output_lock;
+	spinlock_t open_lock;
+
+	unsigned int channel;
+
+	unsigned int midi_mode;
+	int port;
+	int tx_enable, rx_enable;
+	int ipr_tx, ipr_rx;            
+	
+	int input_avail, output_ready;
+	int ack, reset, enter_uart;
+
+	void (*interrupt)(struct snd_ca_midi *midi, unsigned int status);
+	void (*interrupt_enable)(struct snd_ca_midi *midi, int intr);
+	void (*interrupt_disable)(struct snd_ca_midi *midi, int intr);
+
+	unsigned char (*read)(struct snd_ca_midi *midi, int idx);
+	void (*write)(struct snd_ca_midi *midi, int data, int idx);
+
+	/* get info from dev_id */
+	struct snd_card *(*get_dev_id_card)(void *dev_id);
+	int (*get_dev_id_port)(void *dev_id);
+};
+
+int ca_midi_init(void *card, struct snd_ca_midi *midi, int device, char *name);