Linux-2.6.12-rc2
Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.
Let it rip!
diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile
new file mode 100644
index 0000000..89c6cee
--- /dev/null
+++ b/sound/pci/ca0106/Makefile
@@ -0,0 +1,3 @@
+snd-ca0106-objs := ca0106_main.o ca0106_proc.o ca0106_mixer.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 0000000..deb0288
--- /dev/null
+++ b/sound/pci/ca0106/ca0106.h
@@ -0,0 +1,549 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.20
+ *
+ * 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.
+ *
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+/************************************************************************************************/
+/* PCI function 0 registers, address = <val> + PCIBASE0 */
+/************************************************************************************************/
+
+#define 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 DATA 0x04 /* Indexed register set data register */
+ /* DATA[31:0] */
+
+#define 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 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 UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */
+#define 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 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 = ? / 1 = ?
+ * 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 AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
+
+/********************************************************************************************************/
+/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
+/********************************************************************************************************/
+
+/* Initally 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 addresss */
+ /* 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 -> Sheild 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_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 UART_A_DATA 0x6c /* Uart, used in setting sample rates, bits per sample etc. */
+#define UART_A_CMD 0x6d /* Uart, used in setting sample rates, bits per sample etc. */
+#define UART_B_DATA 0x6e /* Uart, Unknown. */
+#define UART_B_CMD 0x6f /* Uart, Unknown. */
+#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_0 0x7c /* I2C Data Port 0. 32 bit */
+#define I2C_1 0x7d /* I2C Data Port 1. 32 bit */
+
+
+#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
+
+typedef struct snd_ca0106_channel ca0106_channel_t;
+typedef struct snd_ca0106 ca0106_t;
+typedef struct snd_ca0106_pcm ca0106_pcm_t;
+
+struct snd_ca0106_channel {
+ ca0106_t *emu;
+ int number;
+ int use;
+ void (*interrupt)(ca0106_t *emu, ca0106_channel_t *channel);
+ ca0106_pcm_t *epcm;
+};
+
+struct snd_ca0106_pcm {
+ ca0106_t *emu;
+ snd_pcm_substream_t *substream;
+ int channel_id;
+ unsigned short running;
+};
+
+// definition of the chip-specific record
+struct snd_ca0106 {
+ snd_card_t *card;
+ struct pci_dev *pci;
+
+ unsigned long port;
+ struct resource *res_port;
+ int irq;
+
+ unsigned int revision; /* chip revision */
+ unsigned int serial; /* serial number */
+ unsigned short model; /* subsystem id */
+
+ spinlock_t emu_lock;
+
+ ac97_t *ac97;
+ snd_pcm_t *pcm;
+
+ ca0106_channel_t playback_channels[4];
+ ca0106_channel_t capture_channels[4];
+ u32 spdif_bits[4]; /* s/pdif out setup */
+ int spdif_enable;
+ int capture_source;
+
+ struct snd_dma_buffer buffer;
+};
+
+int __devinit snd_ca0106_mixer(ca0106_t *emu);
+int __devinit snd_ca0106_proc_init(ca0106_t * emu);
+
+unsigned int snd_ca0106_ptr_read(ca0106_t * emu,
+ unsigned int reg,
+ unsigned int chn);
+
+void snd_ca0106_ptr_write(ca0106_t *emu,
+ unsigned int reg,
+ unsigned int chn,
+ unsigned int data);
+
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
new file mode 100644
index 0000000..82533b4
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -0,0 +1,1283 @@
+/*
+ * 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:
+ * 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
+ *
+ * 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 initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.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>
+
+MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>");
+MODULE_DESCRIPTION("CA0106");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}");
+
+// module parameters (see "Module Parameters")
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+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.");
+
+#include "ca0106.h"
+
+typedef struct {
+ u32 serial;
+ char * name;
+} ca0106_names_t;
+
+static ca0106_names_t ca0106_chip_names[] = {
+ { 0x10021102, "AudigyLS [SB0310]"} ,
+ { 0x10051102, "AudigyLS [SB0310b]"} , /* Unknown AudigyLS that also says SB0310 on it */
+ { 0x10061102, "Live! 7.1 24bit [SB0410]"} , /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
+ { 0x10071102, "Live! 7.1 24bit [SB0413]"} , /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ { 0x10091462, "MSI K8N Diamond MB [SB0438]"}, /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+ { 0, "AudigyLS [Unknown]" }
+};
+
+/* hardware definition */
+static snd_pcm_hardware_t snd_ca0106_playback_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,
+ .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 snd_pcm_hardware_t 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,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = ((65536 - 64) * 8),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 64),
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+unsigned int snd_ca0106_ptr_read(ca0106_t * 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 + PTR);
+ val = inl(emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+void snd_ca0106_ptr_write(ca0106_t *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 + PTR);
+ outl(data, emu->port + DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_ca0106_intr_enable(ca0106_t *emu, unsigned int intrenb)
+{
+ unsigned long flags;
+ unsigned int enable;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ enable = inl(emu->port + INTE) | intrenb;
+ outl(enable, emu->port + INTE);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static void snd_ca0106_pcm_free_substream(snd_pcm_runtime_t *runtime)
+{
+ ca0106_pcm_t *epcm = runtime->private_data;
+
+ if (epcm) {
+ kfree(epcm);
+ }
+}
+
+/* open_playback callback */
+static int snd_ca0106_pcm_open_playback_channel(snd_pcm_substream_t *substream, int channel_id)
+{
+ ca0106_t *chip = snd_pcm_substream_chip(substream);
+ ca0106_channel_t *channel = &(chip->playback_channels[channel_id]);
+ ca0106_pcm_t *epcm;
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int err;
+
+ epcm = kcalloc(1, 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;
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm=epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_playback(snd_pcm_substream_t *substream)
+{
+ ca0106_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *epcm = runtime->private_data;
+ chip->playback_channels[epcm->channel_id].use=0;
+/* FIXME: maybe zero others */
+ return 0;
+}
+
+static int snd_ca0106_pcm_open_playback_front(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_center_lfe(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_unknown(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
+}
+
+static int snd_ca0106_pcm_open_playback_rear(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
+}
+
+/* open_capture callback */
+static int snd_ca0106_pcm_open_capture_channel(snd_pcm_substream_t *substream, int channel_id)
+{
+ ca0106_t *chip = snd_pcm_substream_chip(substream);
+ ca0106_channel_t *channel = &(chip->capture_channels[channel_id]);
+ ca0106_pcm_t *epcm;
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int err;
+
+ epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);
+ if (epcm == NULL) {
+ snd_printk("open_capture_channel: failed epcm alloc\n");
+ 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;
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
+ channel->epcm=epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+ //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
+ if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
+ return err;
+ return 0;
+}
+
+/* close callback */
+static int snd_ca0106_pcm_close_capture(snd_pcm_substream_t *substream)
+{
+ ca0106_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *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(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 0);
+}
+
+static int snd_ca0106_pcm_open_1_capture(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 1);
+}
+
+static int snd_ca0106_pcm_open_2_capture(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 2);
+}
+
+static int snd_ca0106_pcm_open_3_capture(snd_pcm_substream_t *substream)
+{
+ return snd_ca0106_pcm_open_capture_channel(substream, 3);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_playback(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t * hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_playback(snd_pcm_substream_t *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* hw_params callback */
+static int snd_ca0106_pcm_hw_params_capture(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t * hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_ca0106_pcm_hw_free_capture(snd_pcm_substream_t *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/* prepare playback callback */
+static int snd_ca0106_pcm_prepare_playback(snd_pcm_substream_t *substream)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *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;
+
+ //snd_printk("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));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+ /* 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 + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + 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(snd_pcm_substream_t *substream)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *epcm = runtime->private_data;
+ int channel = epcm->channel_id;
+ //printk("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(snd_pcm_substream_t *substream,
+ int cmd)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime;
+ ca0106_pcm_t *epcm;
+ int channel;
+ int result = 0;
+ struct list_head *pos;
+ snd_pcm_substream_t *s;
+ u32 basic = 0;
+ u32 extended = 0;
+ int running=0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running=1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ default:
+ running=0;
+ break;
+ }
+ snd_pcm_group_for_each(pos, substream) {
+ s = snd_pcm_group_substream_entry(pos);
+ runtime = s->runtime;
+ epcm = runtime->private_data;
+ channel = epcm->channel_id;
+ //snd_printk("channel=%d\n",channel);
+ epcm->running = running;
+ basic |= (0x1<<channel);
+ extended |= (0x10<<channel);
+ snd_pcm_trigger_done(s, substream);
+ }
+ //snd_printk("basic=0x%x, extended=0x%x\n",basic, extended);
+
+ 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) | (extended));
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic));
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
+ snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended));
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
+/* trigger_capture callback */
+static int snd_ca0106_pcm_trigger_capture(snd_pcm_substream_t *substream,
+ int cmd)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *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(snd_pcm_substream_t *substream)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0;
+ int channel = epcm->channel_id;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr2+= (ptr4 >> 3) * runtime->period_size;
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size)
+ ptr -= runtime->buffer_size;
+ //printk("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;
+}
+
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_ca0106_pcm_pointer_capture(snd_pcm_substream_t *substream)
+{
+ ca0106_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ca0106_pcm_t *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ int channel = 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;
+ //printk("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 snd_pcm_ops_t snd_ca0106_playback_front_ops = {
+ .open = snd_ca0106_pcm_open_playback_front,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static snd_pcm_ops_t snd_ca0106_capture_0_ops = {
+ .open = snd_ca0106_pcm_open_0_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static snd_pcm_ops_t snd_ca0106_capture_1_ops = {
+ .open = snd_ca0106_pcm_open_1_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static snd_pcm_ops_t snd_ca0106_capture_2_ops = {
+ .open = snd_ca0106_pcm_open_2_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static snd_pcm_ops_t snd_ca0106_capture_3_ops = {
+ .open = snd_ca0106_pcm_open_3_capture,
+ .close = snd_ca0106_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_capture,
+ .hw_free = snd_ca0106_pcm_hw_free_capture,
+ .prepare = snd_ca0106_pcm_prepare_capture,
+ .trigger = snd_ca0106_pcm_trigger_capture,
+ .pointer = snd_ca0106_pcm_pointer_capture,
+};
+
+static snd_pcm_ops_t snd_ca0106_playback_center_lfe_ops = {
+ .open = snd_ca0106_pcm_open_playback_center_lfe,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static snd_pcm_ops_t snd_ca0106_playback_unknown_ops = {
+ .open = snd_ca0106_pcm_open_playback_unknown,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_playback,
+ .prepare = snd_ca0106_pcm_prepare_playback,
+ .trigger = snd_ca0106_pcm_trigger_playback,
+ .pointer = snd_ca0106_pcm_pointer_playback,
+};
+
+static snd_pcm_ops_t snd_ca0106_playback_rear_ops = {
+ .open = snd_ca0106_pcm_open_playback_rear,
+ .close = snd_ca0106_pcm_close_playback,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ca0106_pcm_hw_params_playback,
+ .hw_free = snd_ca0106_pcm_hw_free_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(ac97_t *ac97,
+ unsigned short reg)
+{
+ ca0106_t *emu = ac97->private_data;
+ unsigned long flags;
+ unsigned short val;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ val = inw(emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+ return val;
+}
+
+static void snd_ca0106_ac97_write(ac97_t *ac97,
+ unsigned short reg, unsigned short val)
+{
+ ca0106_t *emu = ac97->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&emu->emu_lock, flags);
+ outb(reg, emu->port + AC97ADDRESS);
+ outw(val, emu->port + AC97DATA);
+ spin_unlock_irqrestore(&emu->emu_lock, flags);
+}
+
+static int snd_ca0106_ac97(ca0106_t *chip)
+{
+ ac97_bus_t *pbus;
+ ac97_template_t ac97;
+ int err;
+ static ac97_bus_ops_t ops = {
+ .write = snd_ca0106_ac97_write,
+ .read = snd_ca0106_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
+ return err;
+ pbus->no_vra = 1; /* we don't need VRA */
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
+}
+
+static int snd_ca0106_free(ca0106_t *chip)
+{
+ if (chip->res_port != NULL) { /* avoid access to already used hardware */
+ // disable interrupts
+ snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+ outl(0, chip->port + 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 + 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
+ */
+ }
+ // release the data
+#if 1
+ if (chip->buffer.area)
+ snd_dma_free_pages(&chip->buffer);
+#endif
+
+ // release the i/o port
+ if (chip->res_port) {
+ release_resource(chip->res_port);
+ kfree_nocheck(chip->res_port);
+ }
+ // release the irq
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+ pci_disable_device(chip->pci);
+ kfree(chip);
+ return 0;
+}
+
+static int snd_ca0106_dev_free(snd_device_t *device)
+{
+ ca0106_t *chip = device->device_data;
+ return snd_ca0106_free(chip);
+}
+
+static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id,
+ struct pt_regs *regs)
+{
+ unsigned int status;
+
+ ca0106_t *chip = dev_id;
+ int i;
+ int mask;
+ unsigned int stat76;
+ ca0106_channel_t *pchannel;
+
+ spin_lock(&chip->emu_lock);
+
+ status = inl(chip->port + IPR);
+
+ // call updater, unlock before it
+ spin_unlock(&chip->emu_lock);
+
+ if (! status)
+ return IRQ_NONE;
+
+ stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
+ //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
+ //snd_printk("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);
+ //printk(KERN_INFO "interrupt [%d] used\n", i);
+ }
+ }
+ //printk(KERN_INFO "channel=%p\n",pchannel);
+ //printk(KERN_INFO "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);
+ //printk(KERN_INFO "interrupt [%d] used\n", i);
+ }
+ }
+ //printk(KERN_INFO "channel=%p\n",pchannel);
+ //printk(KERN_INFO "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);
+ spin_lock(&chip->emu_lock);
+ // acknowledge the interrupt if necessary
+ outl(status, chip->port+IPR);
+
+ spin_unlock(&chip->emu_lock);
+
+ return IRQ_HANDLED;
+}
+
+static void snd_ca0106_pcm_free(snd_pcm_t *pcm)
+{
+ ca0106_t *emu = pcm->private_data;
+ emu->pcm = NULL;
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int __devinit snd_ca0106_pcm(ca0106_t *emu, int device, snd_pcm_t **rpcm)
+{
+ snd_pcm_t *pcm;
+ snd_pcm_substream_t *substream;
+ int err;
+
+ if (rpcm)
+ *rpcm = NULL;
+ if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0)
+ return err;
+
+ pcm->private_data = emu;
+ pcm->private_free = snd_ca0106_pcm_free;
+
+ 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);
+ 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);
+ 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);
+ 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);
+ break;
+ }
+
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strcpy(pcm->name, "CA0106");
+ emu->pcm = pcm;
+
+ for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
+ return err;
+ }
+
+ for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ substream;
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
+ SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(emu->pci),
+ 64*1024, 64*1024)) < 0)
+ return err;
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+
+ return 0;
+}
+
+static int __devinit snd_ca0106_create(snd_card_t *card,
+ struct pci_dev *pci,
+ ca0106_t **rchip)
+{
+ ca0106_t *chip;
+ int err;
+ int ch;
+ static snd_device_ops_t ops = {
+ .dev_free = snd_ca0106_dev_free,
+ };
+
+ *rchip = NULL;
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (pci_set_dma_mask(pci, 0xffffffffUL) < 0 ||
+ pci_set_consistent_dma_mask(pci, 0xffffffffUL) < 0) {
+ printk(KERN_ERR "error to set 32bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ spin_lock_init(&chip->emu_lock);
+
+ chip->port = pci_resource_start(pci, 0);
+ if ((chip->res_port = request_region(chip->port, 0x20,
+ "snd_ca0106")) == NULL) {
+ snd_ca0106_free(chip);
+ printk(KERN_ERR "cannot allocate the port\n");
+ return -EBUSY;
+ }
+
+ if (request_irq(pci->irq, snd_ca0106_interrupt,
+ SA_INTERRUPT|SA_SHIRQ, "snd_ca0106",
+ (void *)chip)) {
+ snd_ca0106_free(chip);
+ printk(KERN_ERR "cannot grab irq\n");
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* This stores the periods table. */
+ if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) {
+ snd_ca0106_free(chip);
+ return -ENOMEM;
+ }
+
+ pci_set_master(pci);
+ /* read revision & serial */
+ pci_read_config_byte(pci, PCI_REVISION_ID, (char *)&chip->revision);
+ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
+ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+#if 1
+ printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model,
+ chip->revision, chip->serial);
+#endif
+
+ outl(0, chip->port + 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)
+ */
+ snd_ca0106_ptr_write(chip, SPCS0, 0,
+ chip->spdif_bits[0] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ /* Only SPCS1 has been tested */
+ snd_ca0106_ptr_write(chip, SPCS1, 0,
+ chip->spdif_bits[1] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_ca0106_ptr_write(chip, SPCS2, 0,
+ chip->spdif_bits[2] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+ snd_ca0106_ptr_write(chip, SPCS3, 0,
+ chip->spdif_bits[3] =
+ SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
+ SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
+ SPCS_GENERATIONSTATUS | 0x00001200 |
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
+
+ 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 + AC97ADDRESS);
+ outw(0x8000, chip->port + AC97DATA);
+#if 0
+ 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
+
+ //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
+ /* Analog or Digital output */
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000b0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers */
+ chip->spdif_enable = 0; /* Set digital SPDIF output off */
+ chip->capture_source = 3; /* Set CAPTURE_SOURCE */
+ //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
+ //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
+
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
+ snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
+ 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++) {
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
+ snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
+ //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */
+ //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
+ snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
+ }
+ snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
+ chip->capture_source = 3; /* Set CAPTURE_SOURCE */
+
+ if ((chip->serial == 0x10061102) ||
+ (chip->serial == 0x10071102) ||
+ (chip->serial == 0x10091462)) { /* 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+GPIO);
+ //outl(0x00f0e000, chip->port+GPIO); /* Analog */
+ outl(0x005f4300, chip->port+GPIO); /* Analog */
+ } else {
+ outl(0x0, chip->port+GPIO);
+ outl(0x005f03a3, chip->port+GPIO); /* Analog */
+ //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */
+ }
+ snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
+
+ //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
+ //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
+ //outl(0x00000009, chip->port+HCFG);
+ outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_ca0106_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_ca0106_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ snd_card_t *card;
+ ca0106_t *chip;
+ ca0106_names_t *c;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ if ((err = snd_ca0106_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ if ((chip->serial != 0x10061102) &&
+ (chip->serial != 0x10071102) &&
+ (chip->serial != 0x10091462) ) { /* The SB0410 and SB0413 do not have an ac97 chip. */
+ if ((err = snd_ca0106_ac97(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ }
+ if ((err = snd_ca0106_mixer(chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ snd_ca0106_proc_init(chip);
+
+ strcpy(card->driver, "CA0106");
+ strcpy(card->shortname, "CA0106");
+
+ for (c=ca0106_chip_names; c->serial; c++) {
+ if (c->serial == chip->serial) break;
+ }
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ c->name, chip->port, chip->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_ca0106_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+// PCI IDs
+static struct pci_device_id snd_ca0106_ids[] = {
+ { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
+
+// pci_driver definition
+static struct pci_driver driver = {
+ .name = "CA0106",
+ .id_table = snd_ca0106_ids,
+ .probe = snd_ca0106_probe,
+ .remove = __devexit_p(snd_ca0106_remove),
+};
+
+// initialization of the module
+static int __init alsa_card_ca0106_init(void)
+{
+ int err;
+
+ if ((err = pci_module_init(&driver)) > 0)
+ return err;
+
+ return 0;
+}
+
+// clean up the module
+static void __exit alsa_card_ca0106_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_ca0106_init)
+module_exit(alsa_card_ca0106_exit)
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
new file mode 100644
index 0000000..97bed1b
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -0,0 +1,634 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.16
+ *
+ * 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.
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.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 "ca0106.h"
+
+static int snd_ca0106_shared_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int snd_ca0106_shared_spdif_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->spdif_enable;
+ return 0;
+}
+
+static int snd_ca0106_shared_spdif_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->spdif_enable != val);
+ if (change) {
+ emu->spdif_enable = val;
+ if (val == 1) {
+ /* Digital */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
+ snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
+ mask = inl(emu->port + GPIO) & ~0x101;
+ outl(mask, emu->port + GPIO);
+
+ } else {
+ /* Analog */
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000b0000);
+ snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
+ snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
+ mask = inl(emu->port + GPIO) | 0x101;
+ outl(mask, emu->port + GPIO);
+ }
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_ca0106_shared_spdif __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Out",
+ .info = snd_ca0106_shared_spdif_info,
+ .get = snd_ca0106_shared_spdif_get,
+ .put = snd_ca0106_shared_spdif_put
+};
+
+static int snd_ca0106_capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[6] = { "SPDIF out", "i2s mixer out", "SPDIF in", "i2s in", "AC97 in", "SRC out" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 6;
+ if (uinfo->value.enumerated.item > 5)
+ uinfo->value.enumerated.item = 5;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_capture_source_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_source;
+ return 0;
+}
+
+static int snd_ca0106_capture_source_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+ u32 source;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->capture_source != val);
+ if (change) {
+ emu->capture_source = val;
+ 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);
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_ca0106_capture_source __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_ca0106_capture_source_info,
+ .get = snd_ca0106_capture_source_get,
+ .put = snd_ca0106_capture_source_put
+};
+
+static int snd_ca0106_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_ca0106_spdif_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
+ return 0;
+}
+
+static int snd_ca0106_spdif_get_mask(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * 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 int snd_ca0106_spdif_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ int change;
+ unsigned int val;
+
+ val = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+ change = val != emu->spdif_bits[idx];
+ if (change) {
+ snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val);
+ emu->spdif_bits[idx] = val;
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_ca0106_spdif_mask_control =
+{
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get_mask
+};
+
+static snd_kcontrol_new_t snd_ca0106_spdif_control =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .count = 4,
+ .info = snd_ca0106_spdif_info,
+ .get = snd_ca0106_spdif_get,
+ .put = snd_ca0106_spdif_put
+};
+
+static int snd_ca0106_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * 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(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int value;
+
+ 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_get_spdif_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_ca0106_volume_get_spdif_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_get_spdif_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_get_spdif_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_get_analog_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_ca0106_volume_get_analog_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_get_analog_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_get_analog_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_ca0106_volume_get_feedback(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = 1;
+ int reg = CAPTURE_CONTROL;
+ return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_ca0106_volume_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int value;
+ //value = snd_ca0106_ptr_read(emu, reg, channel_id);
+ //value = value & 0xffff;
+ value = ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16);
+ value = value | ((0xff - ucontrol->value.integer.value[0]) << 8) | ((0xff - ucontrol->value.integer.value[1]) );
+ snd_ca0106_ptr_write(emu, reg, channel_id, value);
+ return 1;
+}
+static int snd_ca0106_volume_put_spdif_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_spdif_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_spdif_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_spdif_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME1;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_analog_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_analog_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_analog_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_ca0106_volume_put_analog_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_ca0106_volume_put_feedback(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = 1;
+ int reg = CAPTURE_CONTROL;
+ return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+
+static snd_kcontrol_new_t snd_ca0106_volume_control_analog_front =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Front Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_analog_front,
+ .put = snd_ca0106_volume_put_analog_front
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_analog_center_lfe =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Center/LFE Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_analog_center_lfe,
+ .put = snd_ca0106_volume_put_analog_center_lfe
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_analog_unknown =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Unknown Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_analog_unknown,
+ .put = snd_ca0106_volume_put_analog_unknown
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_analog_rear =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Rear Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_analog_rear,
+ .put = snd_ca0106_volume_put_analog_rear
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_front =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Front Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_spdif_front,
+ .put = snd_ca0106_volume_put_spdif_front
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_center_lfe =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Center/LFE Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_spdif_center_lfe,
+ .put = snd_ca0106_volume_put_spdif_center_lfe
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_unknown =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Unknown Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_spdif_unknown,
+ .put = snd_ca0106_volume_put_spdif_unknown
+};
+static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_rear =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Rear Volume",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_spdif_rear,
+ .put = snd_ca0106_volume_put_spdif_rear
+};
+
+static snd_kcontrol_new_t snd_ca0106_volume_control_feedback =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CAPTURE feedback into PLAYBACK",
+ .info = snd_ca0106_volume_info,
+ .get = snd_ca0106_volume_get_feedback,
+ .put = snd_ca0106_volume_put_feedback
+};
+
+
+static int remove_ctl(snd_card_t *card, const char *name)
+{
+ snd_ctl_elem_id_t 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 snd_kcontrol_t *ctl_find(snd_card_t *card, const char *name)
+{
+ snd_ctl_elem_id_t sid;
+ memset(&sid, 0, sizeof(sid));
+ /* FIXME: strcpy is bad. */
+ strcpy(sid.name, name);
+ sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ return snd_ctl_find_id(card, &sid);
+}
+
+static int rename_ctl(snd_card_t *card, const char *src, const char *dst)
+{
+ snd_kcontrol_t *kctl = ctl_find(card, src);
+ if (kctl) {
+ strcpy(kctl->id.name, dst);
+ return 0;
+ }
+ return -ENOENT;
+}
+
+int __devinit snd_ca0106_mixer(ca0106_t *emu)
+{
+ int err;
+ snd_kcontrol_t *kctl;
+ snd_card_t *card = emu->card;
+ char **c;
+ static char *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",
+ "PC Speaker Playback Switch",
+ "PC Speaker Playback Volume",
+ "Mono Output Select",
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "External Amplifier",
+ "Sigmatel 4-Speaker Stereo Playback Switch",
+ "Sigmatel Surround Phase Inversion Playback ",
+ NULL
+ };
+ static char *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
+
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_front, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_rear, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_center_lfe, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_unknown, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_front, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_rear, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_center_lfe, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_unknown, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_feedback, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_mask_control, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_shared_spdif, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_ca0106_capture_source, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = ctl_find(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT))) != NULL) {
+ /* already defined by ac97, remove it */
+ /* FIXME: or do we need both controls? */
+ remove_ctl(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT));
+ }
+ if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_control, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ return 0;
+}
+
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
new file mode 100644
index 0000000..afb7114
--- /dev/null
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -0,0 +1,436 @@
+/*
+ * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
+ * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
+ * Version: 0.0.17
+ *
+ * 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.
+ *
+ * This code was initally based on code from ALSA's emu10k1x.c which is:
+ * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.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/asoundef.h>
+
+#include "ca0106.h"
+
+
+struct snd_ca0106_category_str {
+ int val;
+ const char *name;
+};
+
+static 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" },
+};
+
+
+void snd_ca0106_proc_dump_iec958( snd_info_buffer_t *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, "unkown\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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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", ®, &val) != 2)
+ continue;
+ if ((reg < 0x40) && (reg >=0) && (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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *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", ®, &channel_id, &val) != 3)
+ continue;
+ if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) )
+ snd_ca0106_ptr_write(emu, reg, channel_id, val);
+ }
+}
+
+
+int __devinit snd_ca0106_proc_init(ca0106_t * emu)
+{
+ snd_info_entry_t *entry;
+
+ if(! snd_card_proc_new(emu->card, "iec958", &entry))
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) {
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32);
+ entry->c.text.write_size = 64;
+ entry->c.text.write = snd_ca0106_proc_reg_write32;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16);
+ if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry))
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8);
+ if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) {
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1);
+ entry->c.text.write_size = 64;
+ entry->c.text.write = snd_ca0106_proc_reg_write;
+// entry->private_data = emu;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry))
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2);
+ return 0;
+}
+