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Update hdspm.h file from Adrian Knoth
To satisfy new hdspmixer requirements, update hdspm.h file. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This commit is contained in:
Jaroslav Kysela
parent
59977f90b4
commit
f0f8d22701
2 changed files with 308 additions and 43 deletions
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@ -1,6 +1,6 @@
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alsasoundincludedir = ${includedir}/alsa/sound
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alsasoundinclude_HEADERS = asound_fm.h hdsp.h sb16_csp.h sscape_ioctl.h emu10k1.h \
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type_compat.h
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alsasoundinclude_HEADERS = asound_fm.h hdsp.h hdspm.h sb16_csp.h \
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sscape_ioctl.h emu10k1.h type_compat.h
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noinst_HEADERS = asound.h asoundef.h asequencer.h
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@ -23,50 +23,41 @@
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/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
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#define HDSPM_MAX_CHANNELS 64
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enum hdspm_io_type {
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MADI,
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MADIface,
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AIO,
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AES32,
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RayDAT
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};
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enum hdspm_speed {
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ss,
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ds,
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qs
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};
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/* -------------------- IOCTL Peak/RMS Meters -------------------- */
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/* peam rms level structure like we get from hardware
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maybe in future we can memory map it so I just copy it
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to user on ioctl call now an dont change anything
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rms are made out of low and high values
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where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24)
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(i asume so from the code)
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*/
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struct hdspm_peak_rms {
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uint32_t input_peaks[64];
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uint32_t playback_peaks[64];
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uint32_t output_peaks[64];
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unsigned int level_offset[1024];
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uint64_t input_rms[64];
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uint64_t playback_rms[64];
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uint64_t output_rms[64];
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unsigned int input_peak[64];
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unsigned int playback_peak[64];
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unsigned int output_peak[64];
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unsigned int xxx_peak[64]; /* not used */
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unsigned int reserved[256]; /* not used */
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unsigned int input_rms_l[64];
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unsigned int playback_rms_l[64];
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unsigned int output_rms_l[64];
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unsigned int xxx_rms_l[64]; /* not used */
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unsigned int input_rms_h[64];
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unsigned int playback_rms_h[64];
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unsigned int output_rms_h[64];
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unsigned int xxx_rms_h[64]; /* not used */
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uint8_t speed; /* enum {ss, ds, qs} */
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int status2;
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};
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struct hdspm_peak_rms_ioctl {
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struct hdspm_peak_rms *peak;
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};
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/* use indirect access due to the limit of ioctl bit size */
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#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \
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_IOR('H', 0x40, struct hdspm_peak_rms_ioctl)
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_IOR('H', 0x42, struct hdspm_peak_rms)
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/* ------------ CONFIG block IOCTL ---------------------- */
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struct hdspm_config_info {
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struct hdspm_config {
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unsigned char pref_sync_ref;
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unsigned char wordclock_sync_check;
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unsigned char madi_sync_check;
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@ -80,19 +71,122 @@ struct hdspm_config_info {
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unsigned int analog_out;
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};
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#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \
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_IOR('H', 0x41, struct hdspm_config_info)
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#define SNDRV_HDSPM_IOCTL_GET_CONFIG \
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_IOR('H', 0x41, struct hdspm_config)
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/**
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* If there's a TCO (TimeCode Option) board installed,
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* there are further options and status data available.
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* The hdspm_ltc structure contains the current SMPTE
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* timecode and some status information and can be
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* obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the
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* hdspm_status struct.
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**/
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/* get Soundcard Version */
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enum hdspm_ltc_format {
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format_invalid,
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fps_24,
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fps_25,
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fps_2997,
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fps_30
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};
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enum hdspm_ltc_frame {
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frame_invalid,
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drop_frame,
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full_frame
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};
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enum hdspm_ltc_input_format {
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ntsc,
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pal,
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no_video
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};
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struct hdspm_ltc {
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unsigned int ltc;
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enum hdspm_ltc_format format;
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enum hdspm_ltc_frame frame;
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enum hdspm_ltc_input_format input_format;
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};
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#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
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/**
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* The status data reflects the device's current state
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* as determined by the card's configuration and
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* connection status.
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**/
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enum hdspm_sync {
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hdspm_sync_no_lock = 0,
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hdspm_sync_lock = 1,
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hdspm_sync_sync = 2
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};
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enum hdspm_madi_input {
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hdspm_input_optical = 0,
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hdspm_input_coax = 1
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};
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enum hdspm_madi_channel_format {
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hdspm_format_ch_64 = 0,
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hdspm_format_ch_56 = 1
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};
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enum hdspm_madi_frame_format {
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hdspm_frame_48 = 0,
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hdspm_frame_96 = 1
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};
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enum hdspm_syncsource {
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syncsource_wc = 0,
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syncsource_madi = 1,
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syncsource_tco = 2,
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syncsource_sync = 3,
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syncsource_none = 4
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};
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struct hdspm_status {
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uint8_t card_type; /* enum hdspm_io_type */
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enum hdspm_syncsource autosync_source;
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uint64_t card_clock;
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uint32_t master_period;
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union {
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struct {
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uint8_t sync_wc; /* enum hdspm_sync */
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uint8_t sync_madi; /* enum hdspm_sync */
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uint8_t sync_tco; /* enum hdspm_sync */
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uint8_t sync_in; /* enum hdspm_sync */
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uint8_t madi_input; /* enum hdspm_madi_input */
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uint8_t channel_format; /* enum hdspm_madi_channel_format */
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uint8_t frame_format; /* enum hdspm_madi_frame_format */
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} madi;
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} card_specific;
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};
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#define SNDRV_HDSPM_IOCTL_GET_STATUS \
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_IOR('H', 0x47, struct hdspm_status)
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/**
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* Get information about the card and its add-ons.
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**/
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#define HDSPM_ADDON_TCO 1
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struct hdspm_version {
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uint8_t card_type; /* enum hdspm_io_type */
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char cardname[20];
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unsigned int serial;
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unsigned short firmware_rev;
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int addons;
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};
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#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdspm_version)
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/* ------------- get Matrix Mixer IOCTL --------------- */
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/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte =
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@ -131,4 +225,175 @@ typedef struct hdspm_version hdspm_version_t;
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typedef struct hdspm_channelfader snd_hdspm_channelfader_t;
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typedef struct hdspm_mixer hdspm_mixer_t;
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#endif /* __SOUND_HDSPM_H */
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/* These tables map the ALSA channels 1..N to the channels that we
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need to use in order to find the relevant channel buffer. RME
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refers to this kind of mapping as between "the ADAT channel and
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the DMA channel." We index it using the logical audio channel,
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and the value is the DMA channel (i.e. channel buffer number)
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where the data for that channel can be read/written from/to.
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*/
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static char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = {
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0, 1, 2, 3, 4, 5, 6, 7,
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8, 9, 10, 11, 12, 13, 14, 15,
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16, 17, 18, 19, 20, 21, 22, 23,
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24, 25, 26, 27, 28, 29, 30, 31,
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32, 33, 34, 35, 36, 37, 38, 39,
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40, 41, 42, 43, 44, 45, 46, 47,
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48, 49, 50, 51, 52, 53, 54, 55,
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56, 57, 58, 59, 60, 61, 62, 63
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};
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static char channel_map_unity_ds[HDSPM_MAX_CHANNELS] = {
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0, 2, 4, 6, 8, 10, 12, 14,
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16, 18, 20, 22, 24, 26, 28, 30,
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32, 34, 36, 38, 40, 42, 44, 46,
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48, 50, 52, 54, 56, 58, 60, 62,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_unity_qs[HDSPM_MAX_CHANNELS] = {
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0, 4, 8, 12, 16, 20, 24, 28,
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32, 36, 40, 44, 48, 52, 56, 60,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_raydat_ss[HDSPM_MAX_CHANNELS] = {
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4, 5, 6, 7, 8, 9, 10, 11, /* ADAT 1 */
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12, 13, 14, 15, 16, 17, 18, 19, /* ADAT 2 */
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20, 21, 22, 23, 24, 25, 26, 27, /* ADAT 3 */
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28, 29, 30, 31, 32, 33, 34, 35, /* ADAT 4 */
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0, 1, /* AES */
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2, 3, /* SPDIF */
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-1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_raydat_ds[HDSPM_MAX_CHANNELS] = {
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4, 5, 6, 7, /* ADAT 1 */
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8, 9, 10, 11, /* ADAT 2 */
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12, 13, 14, 15, /* ADAT 3 */
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16, 17, 18, 19, /* ADAT 4 */
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0, 1, /* AES */
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2, 3, /* SPDIF */
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-1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = {
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4, 5, /* ADAT 1 */
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6, 7, /* ADAT 2 */
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8, 9, /* ADAT 3 */
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10, 11, /* ADAT 4 */
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0, 1, /* AES */
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2, 3, /* SPDIF */
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-1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line in */
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8, 9, /* aes in, */
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10, 11, /* spdif in */
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12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */
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-1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line out */
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8, 9, /* aes out */
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10, 11, /* spdif out */
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12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */
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6, 7, /* phone out */
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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};
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static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line in */
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8, 9, /* aes in */
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10, 11, /* spdif in */
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12, 14, 16, 18, /* adat in */
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-1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1
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};
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static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line out */
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8, 9, /* aes out */
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10, 11, /* spdif out */
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12, 14, 16, 18, /* adat out */
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6, 7, /* phone out */
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-1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1
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};
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static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line in */
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8, 9, /* aes in */
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10, 11, /* spdif in */
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12, 16, /* adat in */
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1
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};
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static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
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0, 1, /* line out */
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8, 9, /* aes out */
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10, 11, /* spdif out */
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12, 16, /* adat out */
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6, 7, /* phone out */
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-1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1,
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-1, -1, -1, -1, -1, -1, -1, -1
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};
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#endif
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