ALSA: hdspm - Add support for RME RayDAT and AIO

Incorporate changes by Florian Faber into hdspm.c. Code taken from

   http://wiki.linuxproaudio.org/index.php/Driver:hdspe

Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)

The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

include/sound/hdspm.h

@@ -3,8 +3,8 @@
 /*
  *   Copyright (C) 2003 Winfried Ritsch (IEM)
  *   based on hdsp.h from Thomas Charbonnel (thomas@undata.org)
- *                      
- *    
+ *
+ *
  *   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
@@ -23,50 +23,41 @@
 /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
 #define HDSPM_MAX_CHANNELS      64
 
-/* -------------------- IOCTL Peak/RMS Meters -------------------- */
-
-/* peam rms level structure like we get from hardware 
-  
-   maybe in future we can memory map it so I just copy it
-   to user on ioctl call now an dont change anything
-   rms are made out of low and high values
-   where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24)
-   (i asume so from the code)
-*/
-
-struct hdspm_peak_rms {
-
-	unsigned int level_offset[1024];
+enum hdspm_io_type {
+	MADI,
+	MADIface,
+	AIO,
+	AES32,
+	RayDAT
+};
 
-	unsigned int input_peak[64];
-	unsigned int playback_peak[64];
-	unsigned int output_peak[64];
-	unsigned int xxx_peak[64];	/* not used */
+enum hdspm_speed {
+	ss,
+	ds,
+	qs
+};
 
-	unsigned int reserved[256];	/* not used */
+/* -------------------- IOCTL Peak/RMS Meters -------------------- */
 
-	unsigned int input_rms_l[64];
-	unsigned int playback_rms_l[64];
-	unsigned int output_rms_l[64];
-	unsigned int xxx_rms_l[64];	/* not used */
+struct hdspm_peak_rms {
+	uint32_t input_peaks[64];
+	uint32_t playback_peaks[64];
+	uint32_t output_peaks[64];
 
-	unsigned int input_rms_h[64];
-	unsigned int playback_rms_h[64];
-	unsigned int output_rms_h[64];
-	unsigned int xxx_rms_h[64];	/* not used */
-};
+	uint64_t input_rms[64];
+	uint64_t playback_rms[64];
+	uint64_t output_rms[64];
 
-struct hdspm_peak_rms_ioctl {
-	struct hdspm_peak_rms *peak;
+	uint8_t speed; /* enum {ss, ds, qs} */
+	int status2;
 };
 
-/* use indirect access due to the limit of ioctl bit size */
 #define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \
-	_IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
+	_IOR('H', 0x42, struct hdspm_peak_rms)
 
 /* ------------ CONFIG block IOCTL ---------------------- */
 
-struct hdspm_config_info {
+struct hdspm_config {
 	unsigned char pref_sync_ref;
 	unsigned char wordclock_sync_check;
 	unsigned char madi_sync_check;
@@ -80,18 +71,121 @@ struct hdspm_config_info {
 	unsigned int analog_out;
 };
 
-#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \
-	_IOR('H', 0x41, struct hdspm_config_info)
+#define SNDRV_HDSPM_IOCTL_GET_CONFIG \
+	_IOR('H', 0x41, struct hdspm_config)
+
+/**
+ * If there's a TCO (TimeCode Option) board installed,
+ * there are further options and status data available.
+ * The hdspm_ltc structure contains the current SMPTE
+ * timecode and some status information and can be
+ * obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the
+ * hdspm_status struct.
+ **/
+
+enum hdspm_ltc_format {
+	format_invalid,
+	fps_24,
+	fps_25,
+	fps_2997,
+	fps_30
+};
+
+enum hdspm_ltc_frame {
+	frame_invalid,
+	drop_frame,
+	full_frame
+};
+
+enum hdspm_ltc_input_format {
+	ntsc,
+	pal,
+	no_video
+};
+
+struct hdspm_ltc {
+	unsigned int ltc;
+
+	enum hdspm_ltc_format format;
+	enum hdspm_ltc_frame frame;
+	enum hdspm_ltc_input_format input_format;
+};
+
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+
+/**
+ * The status data reflects the device's current state
+ * as determined by the card's configuration and
+ * connection status.
+ **/
+
+enum hdspm_sync {
+	hdspm_sync_no_lock = 0,
+	hdspm_sync_lock = 1,
+	hdspm_sync_sync = 2
+};
+
+enum hdspm_madi_input {
+	hdspm_input_optical = 0,
+	hdspm_input_coax = 1
+};
+
+enum hdspm_madi_channel_format {
+	hdspm_format_ch_64 = 0,
+	hdspm_format_ch_56 = 1
+};
+
+enum hdspm_madi_frame_format {
+	hdspm_frame_48 = 0,
+	hdspm_frame_96 = 1
+};
+
+enum hdspm_syncsource {
+	syncsource_wc = 0,
+	syncsource_madi = 1,
+	syncsource_tco = 2,
+	syncsource_sync = 3,
+	syncsource_none = 4
+};
+
+struct hdspm_status {
+	uint8_t card_type; /* enum hdspm_io_type */
+	enum hdspm_syncsource autosync_source;
 
+	uint64_t card_clock;
+	uint32_t master_period;
+
+	union {
+		struct {
+			uint8_t sync_wc; /* enum hdspm_sync */
+			uint8_t sync_madi; /* enum hdspm_sync */
+			uint8_t sync_tco; /* enum hdspm_sync */
+			uint8_t sync_in; /* enum hdspm_sync */
+			uint8_t madi_input; /* enum hdspm_madi_input */
+			uint8_t channel_format; /* enum hdspm_madi_channel_format */
+			uint8_t frame_format; /* enum hdspm_madi_frame_format */
+		} madi;
+	} card_specific;
+};
 
-/* get Soundcard Version */
+#define SNDRV_HDSPM_IOCTL_GET_STATUS \
+	_IOR('H', 0x47, struct hdspm_status)
+
+/**
+ * Get information about the card and its add-ons.
+ **/
+
+#define HDSPM_ADDON_TCO 1
 
 struct hdspm_version {
+	uint8_t card_type; /* enum hdspm_io_type */
+	char cardname[20];
+	unsigned int serial;
 	unsigned short firmware_rev;
+	int addons;
 };
 
-#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdspm_version)
-
+#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x48, struct hdspm_version)
 
 /* ------------- get Matrix Mixer IOCTL --------------- */
 
@@ -103,7 +197,7 @@ struct hdspm_version {
 /* equivalent to hardware definition, maybe for future feature of mmap of
  * them
  */
-/* each of 64 outputs has 64 infader and 64 outfader: 
+/* each of 64 outputs has 64 infader and 64 outfader:
    Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */
 
 #define HDSPM_MIXER_CHANNELS HDSPM_MAX_CHANNELS
@@ -131,4 +225,175 @@ typedef struct hdspm_version hdspm_version_t;
 typedef struct hdspm_channelfader snd_hdspm_channelfader_t;
 typedef struct hdspm_mixer hdspm_mixer_t;
 
-#endif				/* __SOUND_HDSPM_H */
+/* These tables map the ALSA channels 1..N to the channels that we
+   need to use in order to find the relevant channel buffer. RME
+   refers to this kind of mapping as between "the ADAT channel and
+   the DMA channel." We index it using the logical audio channel,
+   and the value is the DMA channel (i.e. channel buffer number)
+   where the data for that channel can be read/written from/to.
+*/
+
+char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = {
+	0, 1, 2, 3, 4, 5, 6, 7,
+	8, 9, 10, 11, 12, 13, 14, 15,
+	16, 17, 18, 19, 20, 21, 22, 23,
+	24, 25, 26, 27, 28, 29, 30, 31,
+	32, 33, 34, 35, 36, 37, 38, 39,
+	40, 41, 42, 43, 44, 45, 46, 47,
+	48, 49, 50, 51, 52, 53, 54, 55,
+	56, 57, 58, 59, 60, 61, 62, 63
+};
+
+char channel_map_unity_ds[HDSPM_MAX_CHANNELS] = {
+	0, 2, 4, 6, 8, 10, 12, 14,
+	16, 18, 20, 22, 24, 26, 28, 30,
+	32, 34, 36, 38, 40, 42, 44, 46,
+	48, 50, 52, 54, 56, 58, 60, 62,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_unity_qs[HDSPM_MAX_CHANNELS] = {
+	0, 4, 8, 12, 16, 20, 24, 28,
+	32, 36, 40, 44, 48, 52, 56, 60,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_raydat_ss[HDSPM_MAX_CHANNELS] = {
+	4, 5, 6, 7, 8, 9, 10, 11,	/* ADAT 1 */
+	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT 2 */
+	20, 21, 22, 23, 24, 25, 26, 27,	/* ADAT 3 */
+	28, 29, 30, 31, 32, 33, 34, 35,	/* ADAT 4 */
+	0, 1,			/* AES */
+	2, 3,			/* SPDIF */
+	-1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_raydat_ds[HDSPM_MAX_CHANNELS] = {
+	4, 5, 6, 7,		/* ADAT 1 */
+	8, 9, 10, 11,		/* ADAT 2 */
+	12, 13, 14, 15,		/* ADAT 3 */
+	16, 17, 18, 19,		/* ADAT 4 */
+	0, 1,			/* AES */
+	2, 3,			/* SPDIF */
+	-1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = {
+	4, 5,			/* ADAT 1 */
+	6, 7,			/* ADAT 2 */
+	8, 9,			/* ADAT 3 */
+	10, 11,			/* ADAT 4 */
+	0, 1,			/* AES */
+	2, 3,			/* SPDIF */
+	-1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line in */
+	8, 9,			/* aes in, */
+	10, 11,			/* spdif in */
+	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT in */
+	-1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line out */
+	8, 9,			/* aes out */
+	10, 11,			/* spdif out */
+	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT out */
+	6, 7,			/* phone out */
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+};
+
+char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line in */
+	8, 9,			/* aes in */
+	10, 11,			/* spdif in */
+	12, 14, 16, 18,		/* adat in */
+	-1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1
+};
+
+char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line out */
+	8, 9,			/* aes out */
+	10, 11,			/* spdif out */
+	12, 14, 16, 18,		/* adat out */
+	6, 7,			/* phone out */
+	-1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1
+};
+
+char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line in */
+	8, 9,			/* aes in */
+	10, 11,			/* spdif in */
+	12, 16,			/* adat in */
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1
+};
+
+char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
+	0, 1,			/* line out */
+	8, 9,			/* aes out */
+	10, 11,			/* spdif out */
+	12, 16,			/* adat out */
+	6, 7,			/* phone out */
+	-1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1,
+	-1, -1, -1, -1, -1, -1, -1, -1
+};
+
+#endif