Merged the changes by Florian Faber to the hdspm driver. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Whitespace fixes, also added missing commas to the texts_ports_* arrays. Changed card_clock to unsigned long long in struct hdspm_status.
Signed-off-by: Adrian Knoth adi@drcomp.erfurt.thur.de
diff --git a/include/hdspm.h b/include/hdspm.h index 81990b2..5b69694 100644 --- a/include/hdspm.h +++ b/include/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 +23,37 @@ /* 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 { +typedef enum { + MADI, + MADIface, + AIO, + AES32, + RayDAT +} hdspm_io_type;
- unsigned int level_offset[1024];
- unsigned int input_peak[64]; - unsigned int playback_peak[64]; - unsigned int output_peak[64]; - unsigned int xxx_peak[64]; /* not used */ +/* -------------------- IOCTL Peak/RMS Meters -------------------- */
- unsigned int reserved[256]; /* 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_l[64]; - unsigned int playback_rms_l[64]; - unsigned int output_rms_l[64]; - unsigned int xxx_rms_l[64]; /* not used */ + uint64_t input_rms[64]; + uint64_t playback_rms[64]; + uint64_t output_rms[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 */ + enum {ss, ds, qs} speed; + int status2; };
-struct hdspm_peak_rms_ioctl { - struct hdspm_peak_rms *peak; -};
-/* 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; @@ -80,14 +67,118 @@ struct hdspm_config_info { unsigned int analog_out; };
-#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \ - _IOR('H', 0x41, struct hdspm_config_info)
-/* get Soundcard Version */ +#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 + * optained via SNDRV_HDSPM_IOCTL_GET_LTC or in the + * hdspm_status struct. + **/ + +typedef enum { + format_invalid, + fps_24, + fps_25, + fps_2997, + fps_30 +} hdspm_ltc_format; + +typedef enum { + frame_invalid, + drop_frame, + full_frame +} hdspm_ltc_frame; + +typedef enum { + ntsc, + pal, + no_video +} hdspm_ltc_input_format; + +struct hdspm_ltc { + unsigned int ltc; + + hdspm_ltc_format format; + hdspm_ltc_frame frame; + 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. + **/ + +typedef enum { + hdspm_sync_no_lock = 0, + hdspm_sync_lock = 1, + hdspm_sync_sync = 2 +} hdspm_sync; + +typedef enum {hdspm_input_optical = 0, hdspm_input_coax = 1} hdspm_madi_input; + +typedef enum {hdspm_format_ch_64 = 0, + hdspm_format_ch_56 = 1 +} hdspm_madi_channel_format; + +typedef enum {hdspm_frame_48 = 0, + hdspm_frame_96 = 1 +} hdspm_madi_frame_format; + +typedef enum {syncsource_wc = 0, + syncsource_madi = 1, + syncsource_tco = 2, + syncsource_sync = 3, + syncsource_none = 4 +} hdspm_syncsource; + +struct hdspm_status { + hdspm_io_type card_type; + hdspm_syncsource autosync_source; + + unsigned long long card_clock; + unsigned int master_period; + + union { + struct { + hdspm_sync sync_wc; + hdspm_sync sync_madi; + hdspm_sync sync_tco; + hdspm_sync sync_in; + hdspm_madi_input madi_input; + hdspm_madi_channel_format channel_format; + hdspm_madi_frame_format frame_format; + } madi; + } card_specific; +}; + +#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 { + hdspm_io_type card_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) @@ -103,7 +194,7 @@ 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 +222,277 @@ 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 */ + +static char *texts_sync_status[] = { + "no lock", + "lock", + "sync" +}; + +static char *texts_ports_madi[] = { + "MADI.1", "MADI.2", "MADI.3", "MADI.4", "MADI.5", "MADI.6", + "MADI.7", "MADI.8", "MADI.9", "MADI.10", "MADI.11", "MADI.12", + "MADI.13", "MADI.14", "MADI.15", "MADI.16", "MADI.17", "MADI.18", + "MADI.19", "MADI.20", "MADI.21", "MADI.22", "MADI.23", "MADI.24", + "MADI.25", "MADI.26", "MADI.27", "MADI.28", "MADI.29", "MADI.30", + "MADI.31", "MADI.32", "MADI.33", "MADI.34", "MADI.35", "MADI.36", + "MADI.37", "MADI.38", "MADI.39", "MADI.40", "MADI.41", "MADI.42", + "MADI.43", "MADI.44", "MADI.45", "MADI.46", "MADI.47", "MADI.48", + "MADI.49", "MADI.50", "MADI.51", "MADI.52", "MADI.53", "MADI.54", + "MADI.55", "MADI.56", "MADI.57", "MADI.58", "MADI.59", "MADI.60", + "MADI.61", "MADI.62", "MADI.63", "MADI.64", +}; + + +static char *texts_ports_raydat_ss[] = { + "ADAT1.1", "ADAT1.2", "ADAT1.3", "ADAT1.4", "ADAT1.5", "ADAT1.6", + "ADAT1.7", "ADAT1.8", "ADAT2.1", "ADAT2.2", "ADAT2.3", "ADAT2.4", + "ADAT2.5", "ADAT2.6", "ADAT2.7", "ADAT2.8", "ADAT3.1", "ADAT3.2", + "ADAT3.3", "ADAT3.4", "ADAT3.5", "ADAT3.6", "ADAT3.7", "ADAT3.8", + "ADAT4.1", "ADAT4.2", "ADAT4.3", "ADAT4.4", "ADAT4.5", "ADAT4.6", + "ADAT4.7", "ADAT4.8", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R" +}; + +static char *texts_ports_raydat_ds[] = { + "ADAT1.1", "ADAT1.2", "ADAT1.3", "ADAT1.4", + "ADAT2.1", "ADAT2.2", "ADAT2.3", "ADAT2.4", + "ADAT3.1", "ADAT3.2", "ADAT3.3", "ADAT3.4", + "ADAT4.1", "ADAT4.2", "ADAT4.3", "ADAT4.4", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R" +}; + +static char *texts_ports_raydat_qs[] = { + "ADAT1.1", "ADAT1.2", + "ADAT2.1", "ADAT2.2", + "ADAT3.1", "ADAT3.2", + "ADAT4.1", "ADAT4.2", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R" +}; + + +static char *texts_ports_aio_in_ss[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", + "ADAT.7", "ADAT.8" +}; + +static char *texts_ports_aio_out_ss[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", + "ADAT.7", "ADAT.8", + "Phone.L", "Phone.R" +}; + +static char *texts_ports_aio_in_ds[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" +}; + +static char *texts_ports_aio_out_ds[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "Phone.L", "Phone.R" +}; + +static char *texts_ports_aio_in_qs[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" +}; + +static char *texts_ports_aio_out_qs[] = { + "Analogue.L", "Analogue.R", + "AES.L", "AES.R", + "SPDIF.L", "SPDIF.R", + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "Phone.L", "Phone.R" +}; + + +/* 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. +*/ + +static 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 +}; + +static 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,}; + +static 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,}; + + +static 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, +}; + +static 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, +}; + +static 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, +}; + + +static 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, +}; + +static 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, +}; + +static 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 +}; + +static 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 +}; + + +static 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 +}; + +static 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