From 0dca1793063c28dde8f6c49c9c72203fe5cb6efc Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 26 Jan 2011 19:32:14 +0100 Subject: 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 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/hdspm.h | 349 ++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 307 insertions(+), 42 deletions(-) (limited to 'include/sound') diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index 81990b2bcc9..c3f18194b08 100644 --- a/include/sound/hdspm.h +++ b/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 -- cgit v1.2.3 From 55a57606b26665870f2993dc53a43daad157dbcd Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Thu, 27 Jan 2011 11:23:15 +0100 Subject: ALSA: [hdspm] Move static mapping arrays to .c As requested by Takashi and Jaroslav, these arrays should not be in the header file. Signed-off-by: Adrian Knoth Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/hdspm.h | 170 --------------------------------------------- sound/pci/rme9652/hdspm.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 171 insertions(+), 170 deletions(-) (limited to 'include/sound') diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index c3f18194b08..1774ff5ff63 100644 --- a/include/sound/hdspm.h +++ b/include/sound/hdspm.h @@ -225,175 +225,5 @@ typedef struct hdspm_version hdspm_version_t; typedef struct hdspm_channelfader snd_hdspm_channelfader_t; typedef struct hdspm_mixer hdspm_mixer_t; -/* 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 2db871d9a00..28a1eb3f4d0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -673,6 +673,177 @@ static char *texts_ports_aio_out_qs[] = { "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 +}; + struct hdspm_midi { struct hdspm *hdspm; int id; -- cgit v1.2.3 From ea18e137baf3e3e9212bfd7b071555fc712159b5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 26 Jan 2011 11:04:08 +0100 Subject: ALSA: Release v1.0.24 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/version.h b/include/sound/version.h index bf69a5b7e65..8fc5321e1ec 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.23" +#define CONFIG_SND_VERSION "1.0.24" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From fea952e5cc23ea94b4677ca20774cdc3cea014e2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Feb 2011 11:00:47 +0100 Subject: ALSA: core: sparse cleanups Change the core code where sparse complains. In most cases, this means just adding annotations to confirm that we indeed want to do the dirty things we're doing. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/mixer_oss.h | 3 ++ include/sound/pcm.h | 91 +++++++++++++++++++++--------------------- sound/core/device.c | 7 ++-- sound/core/memalloc.c | 3 +- sound/core/oss/linear.c | 7 ++-- sound/core/oss/mixer_oss.c | 10 +++-- sound/core/oss/mulaw.c | 2 +- sound/core/oss/pcm_oss.c | 51 ++++++++++++----------- sound/core/oss/pcm_plugin.c | 40 ++++++++++--------- sound/core/oss/pcm_plugin.h | 11 ++--- sound/core/oss/route.c | 6 +-- sound/core/pcm.c | 10 ++--- sound/core/pcm_misc.c | 35 ++++++++-------- sound/core/pcm_native.c | 2 +- sound/core/seq/seq_clientmgr.c | 7 +--- sound/core/seq/seq_memory.c | 6 +-- sound/core/seq/seq_memory.h | 4 ++ sound/core/vmaster.c | 2 +- 18 files changed, 160 insertions(+), 137 deletions(-) (limited to 'include/sound') diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index 51fbcb4a277..13cb0b430a1 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -73,6 +73,9 @@ struct snd_mixer_oss_file { struct snd_mixer_oss *mixer; }; +int snd_mixer_oss_ioctl_card(struct snd_card *card, + unsigned int cmd, unsigned long arg); + #endif /* CONFIG_SND_MIXER_OSS */ #endif /* __SOUND_MIXER_OSS_H */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e731f8d7193..430a9cc045e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -136,48 +136,49 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) -#define SNDRV_PCM_FMTBIT_S8 (1ULL << SNDRV_PCM_FORMAT_S8) -#define SNDRV_PCM_FMTBIT_U8 (1ULL << SNDRV_PCM_FORMAT_U8) -#define SNDRV_PCM_FMTBIT_S16_LE (1ULL << SNDRV_PCM_FORMAT_S16_LE) -#define SNDRV_PCM_FMTBIT_S16_BE (1ULL << SNDRV_PCM_FORMAT_S16_BE) -#define SNDRV_PCM_FMTBIT_U16_LE (1ULL << SNDRV_PCM_FORMAT_U16_LE) -#define SNDRV_PCM_FMTBIT_U16_BE (1ULL << SNDRV_PCM_FORMAT_U16_BE) -#define SNDRV_PCM_FMTBIT_S24_LE (1ULL << SNDRV_PCM_FORMAT_S24_LE) -#define SNDRV_PCM_FMTBIT_S24_BE (1ULL << SNDRV_PCM_FORMAT_S24_BE) -#define SNDRV_PCM_FMTBIT_U24_LE (1ULL << SNDRV_PCM_FORMAT_U24_LE) -#define SNDRV_PCM_FMTBIT_U24_BE (1ULL << SNDRV_PCM_FORMAT_U24_BE) -#define SNDRV_PCM_FMTBIT_S32_LE (1ULL << SNDRV_PCM_FORMAT_S32_LE) -#define SNDRV_PCM_FMTBIT_S32_BE (1ULL << SNDRV_PCM_FORMAT_S32_BE) -#define SNDRV_PCM_FMTBIT_U32_LE (1ULL << SNDRV_PCM_FORMAT_U32_LE) -#define SNDRV_PCM_FMTBIT_U32_BE (1ULL << SNDRV_PCM_FORMAT_U32_BE) -#define SNDRV_PCM_FMTBIT_FLOAT_LE (1ULL << SNDRV_PCM_FORMAT_FLOAT_LE) -#define SNDRV_PCM_FMTBIT_FLOAT_BE (1ULL << SNDRV_PCM_FORMAT_FLOAT_BE) -#define SNDRV_PCM_FMTBIT_FLOAT64_LE (1ULL << SNDRV_PCM_FORMAT_FLOAT64_LE) -#define SNDRV_PCM_FMTBIT_FLOAT64_BE (1ULL << SNDRV_PCM_FORMAT_FLOAT64_BE) -#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE (1ULL << SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE) -#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE (1ULL << SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE) -#define SNDRV_PCM_FMTBIT_MU_LAW (1ULL << SNDRV_PCM_FORMAT_MU_LAW) -#define SNDRV_PCM_FMTBIT_A_LAW (1ULL << SNDRV_PCM_FORMAT_A_LAW) -#define SNDRV_PCM_FMTBIT_IMA_ADPCM (1ULL << SNDRV_PCM_FORMAT_IMA_ADPCM) -#define SNDRV_PCM_FMTBIT_MPEG (1ULL << SNDRV_PCM_FORMAT_MPEG) -#define SNDRV_PCM_FMTBIT_GSM (1ULL << SNDRV_PCM_FORMAT_GSM) -#define SNDRV_PCM_FMTBIT_SPECIAL (1ULL << SNDRV_PCM_FORMAT_SPECIAL) -#define SNDRV_PCM_FMTBIT_S24_3LE (1ULL << SNDRV_PCM_FORMAT_S24_3LE) -#define SNDRV_PCM_FMTBIT_U24_3LE (1ULL << SNDRV_PCM_FORMAT_U24_3LE) -#define SNDRV_PCM_FMTBIT_S24_3BE (1ULL << SNDRV_PCM_FORMAT_S24_3BE) -#define SNDRV_PCM_FMTBIT_U24_3BE (1ULL << SNDRV_PCM_FORMAT_U24_3BE) -#define SNDRV_PCM_FMTBIT_S20_3LE (1ULL << SNDRV_PCM_FORMAT_S20_3LE) -#define SNDRV_PCM_FMTBIT_U20_3LE (1ULL << SNDRV_PCM_FORMAT_U20_3LE) -#define SNDRV_PCM_FMTBIT_S20_3BE (1ULL << SNDRV_PCM_FORMAT_S20_3BE) -#define SNDRV_PCM_FMTBIT_U20_3BE (1ULL << SNDRV_PCM_FORMAT_U20_3BE) -#define SNDRV_PCM_FMTBIT_S18_3LE (1ULL << SNDRV_PCM_FORMAT_S18_3LE) -#define SNDRV_PCM_FMTBIT_U18_3LE (1ULL << SNDRV_PCM_FORMAT_U18_3LE) -#define SNDRV_PCM_FMTBIT_S18_3BE (1ULL << SNDRV_PCM_FORMAT_S18_3BE) -#define SNDRV_PCM_FMTBIT_U18_3BE (1ULL << SNDRV_PCM_FORMAT_U18_3BE) -#define SNDRV_PCM_FMTBIT_G723_24 (1ULL << SNDRV_PCM_FORMAT_G723_24) -#define SNDRV_PCM_FMTBIT_G723_24_1B (1ULL << SNDRV_PCM_FORMAT_G723_24_1B) -#define SNDRV_PCM_FMTBIT_G723_40 (1ULL << SNDRV_PCM_FORMAT_G723_40) -#define SNDRV_PCM_FMTBIT_G723_40_1B (1ULL << SNDRV_PCM_FORMAT_G723_40_1B) +#define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) +#define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) +#define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) +#define SNDRV_PCM_FMTBIT_S16_LE _SNDRV_PCM_FMTBIT(S16_LE) +#define SNDRV_PCM_FMTBIT_S16_BE _SNDRV_PCM_FMTBIT(S16_BE) +#define SNDRV_PCM_FMTBIT_U16_LE _SNDRV_PCM_FMTBIT(U16_LE) +#define SNDRV_PCM_FMTBIT_U16_BE _SNDRV_PCM_FMTBIT(U16_BE) +#define SNDRV_PCM_FMTBIT_S24_LE _SNDRV_PCM_FMTBIT(S24_LE) +#define SNDRV_PCM_FMTBIT_S24_BE _SNDRV_PCM_FMTBIT(S24_BE) +#define SNDRV_PCM_FMTBIT_U24_LE _SNDRV_PCM_FMTBIT(U24_LE) +#define SNDRV_PCM_FMTBIT_U24_BE _SNDRV_PCM_FMTBIT(U24_BE) +#define SNDRV_PCM_FMTBIT_S32_LE _SNDRV_PCM_FMTBIT(S32_LE) +#define SNDRV_PCM_FMTBIT_S32_BE _SNDRV_PCM_FMTBIT(S32_BE) +#define SNDRV_PCM_FMTBIT_U32_LE _SNDRV_PCM_FMTBIT(U32_LE) +#define SNDRV_PCM_FMTBIT_U32_BE _SNDRV_PCM_FMTBIT(U32_BE) +#define SNDRV_PCM_FMTBIT_FLOAT_LE _SNDRV_PCM_FMTBIT(FLOAT_LE) +#define SNDRV_PCM_FMTBIT_FLOAT_BE _SNDRV_PCM_FMTBIT(FLOAT_BE) +#define SNDRV_PCM_FMTBIT_FLOAT64_LE _SNDRV_PCM_FMTBIT(FLOAT64_LE) +#define SNDRV_PCM_FMTBIT_FLOAT64_BE _SNDRV_PCM_FMTBIT(FLOAT64_BE) +#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE _SNDRV_PCM_FMTBIT(IEC958_SUBFRAME_LE) +#define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE _SNDRV_PCM_FMTBIT(IEC958_SUBFRAME_BE) +#define SNDRV_PCM_FMTBIT_MU_LAW _SNDRV_PCM_FMTBIT(MU_LAW) +#define SNDRV_PCM_FMTBIT_A_LAW _SNDRV_PCM_FMTBIT(A_LAW) +#define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) +#define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) +#define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) +#define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) +#define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) +#define SNDRV_PCM_FMTBIT_S24_3BE _SNDRV_PCM_FMTBIT(S24_3BE) +#define SNDRV_PCM_FMTBIT_U24_3BE _SNDRV_PCM_FMTBIT(U24_3BE) +#define SNDRV_PCM_FMTBIT_S20_3LE _SNDRV_PCM_FMTBIT(S20_3LE) +#define SNDRV_PCM_FMTBIT_U20_3LE _SNDRV_PCM_FMTBIT(U20_3LE) +#define SNDRV_PCM_FMTBIT_S20_3BE _SNDRV_PCM_FMTBIT(S20_3BE) +#define SNDRV_PCM_FMTBIT_U20_3BE _SNDRV_PCM_FMTBIT(U20_3BE) +#define SNDRV_PCM_FMTBIT_S18_3LE _SNDRV_PCM_FMTBIT(S18_3LE) +#define SNDRV_PCM_FMTBIT_U18_3LE _SNDRV_PCM_FMTBIT(U18_3LE) +#define SNDRV_PCM_FMTBIT_S18_3BE _SNDRV_PCM_FMTBIT(S18_3BE) +#define SNDRV_PCM_FMTBIT_U18_3BE _SNDRV_PCM_FMTBIT(U18_3BE) +#define SNDRV_PCM_FMTBIT_G723_24 _SNDRV_PCM_FMTBIT(G723_24) +#define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) +#define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) +#define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE @@ -490,7 +491,7 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, int snd_pcm_status(struct snd_pcm_substream *substream, struct snd_pcm_status *status); int snd_pcm_start(struct snd_pcm_substream *substream); -int snd_pcm_stop(struct snd_pcm_substream *substream, int status); +int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); @@ -748,8 +749,8 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } -#define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)) -#define params_format(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_FORMAT)) +#define params_access(p) ((__force snd_pcm_access_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS))) +#define params_format(p) ((__force snd_pcm_format_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_FORMAT))) #define params_subformat(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_SUBFORMAT)) #define params_channels(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min #define params_rate(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_RATE)->min diff --git a/sound/core/device.c b/sound/core/device.c index a67dfac08c0..2d1ad4b0cd6 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -225,15 +225,16 @@ int snd_device_free_all(struct snd_card *card, snd_device_cmd_t cmd) { struct snd_device *dev; int err; - unsigned int range_low, range_high; + unsigned int range_low, range_high, type; if (snd_BUG_ON(!card)) return -ENXIO; - range_low = cmd * SNDRV_DEV_TYPE_RANGE_SIZE; + range_low = (__force unsigned int)cmd * SNDRV_DEV_TYPE_RANGE_SIZE; range_high = range_low + SNDRV_DEV_TYPE_RANGE_SIZE - 1; __again: list_for_each_entry(dev, &card->devices, list) { - if (dev->type >= range_low && dev->type <= range_high) { + type = (__force unsigned int)dev->type; + if (type >= range_low && type <= range_high) { if ((err = snd_device_free(card, dev->device_data)) < 0) return err; goto __again; diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9e92441f9b7..16bd9c03679 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -192,7 +192,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->bytes = 0; switch (type) { case SNDRV_DMA_TYPE_CONTINUOUS: - dmab->area = snd_malloc_pages(size, (unsigned long)device); + dmab->area = snd_malloc_pages(size, + (__force gfp_t)(unsigned long)device); dmab->addr = 0; break; #ifdef CONFIG_HAS_DMA diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 4c1d1682719..13b3f6f49fa 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -114,7 +114,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, return frames; } -static void init_data(struct linear_priv *data, int src_format, int dst_format) +static void init_data(struct linear_priv *data, + snd_pcm_format_t src_format, snd_pcm_format_t dst_format) { int src_le, dst_le, src_bytes, dst_bytes; @@ -140,9 +141,9 @@ static void init_data(struct linear_priv *data, int src_format, int dst_format) if (snd_pcm_format_signed(src_format) != snd_pcm_format_signed(dst_format)) { if (dst_le) - data->flip = cpu_to_le32(0x80000000); + data->flip = (__force u32)cpu_to_le32(0x80000000); else - data->flip = cpu_to_be32(0x80000000); + data->flip = (__force u32)cpu_to_be32(0x80000000); } } diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 822dd56993c..d8359cfeca1 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -190,9 +190,10 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) return -EIO; if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ int err; - if ((err = mixer->get_recsrc(fmixer, &result)) < 0) + unsigned int index; + if ((err = mixer->get_recsrc(fmixer, &index)) < 0) return err; - result = 1 << result; + result = 1 << index; } else { struct snd_mixer_oss_slot *pslot; int chn; @@ -214,6 +215,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr struct snd_mixer_oss *mixer = fmixer->mixer; struct snd_mixer_oss_slot *pslot; int chn, active; + unsigned int index; int result = 0; if (mixer == NULL) @@ -222,8 +224,8 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr if (recsrc & ~mixer->oss_recsrc) recsrc &= ~mixer->oss_recsrc; mixer->put_recsrc(fmixer, ffz(~recsrc)); - mixer->get_recsrc(fmixer, &result); - result = 1 << result; + mixer->get_recsrc(fmixer, &index); + result = 1 << index; } for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index f7649d4d950..7915564bd39 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -274,7 +274,7 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin, return frames; } -static void init_data(struct mulaw_priv *data, int format) +static void init_data(struct mulaw_priv *data, snd_pcm_format_t format) { #ifdef SNDRV_LITTLE_ENDIAN data->cvt_endian = snd_pcm_format_big_endian(format) > 0; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a2e4eb32469..23c34a02894 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -41,6 +41,7 @@ #include #include #include +#include #define OSS_ALSAEMULVER _SIOR ('M', 249, int) @@ -60,7 +61,6 @@ MODULE_PARM_DESC(nonblock_open, "Don't block opening busy PCM devices."); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM); MODULE_ALIAS_SNDRV_MINOR(SNDRV_MINOR_OSS_PCM1); -extern int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg); static int snd_pcm_oss_get_rate(struct snd_pcm_oss_file *pcm_oss_file); static int snd_pcm_oss_get_channels(struct snd_pcm_oss_file *pcm_oss_file); static int snd_pcm_oss_get_format(struct snd_pcm_oss_file *pcm_oss_file); @@ -656,7 +656,7 @@ snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) #define AFMT_AC3 0x00000400 #define AFMT_VORBIS 0x00000800 -static int snd_pcm_oss_format_from(int format) +static snd_pcm_format_t snd_pcm_oss_format_from(int format) { switch (format) { case AFMT_MU_LAW: return SNDRV_PCM_FORMAT_MU_LAW; @@ -680,7 +680,7 @@ static int snd_pcm_oss_format_from(int format) } } -static int snd_pcm_oss_format_to(int format) +static int snd_pcm_oss_format_to(snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_MU_LAW: return AFMT_MU_LAW; @@ -843,7 +843,8 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) size_t oss_frame_size; int err; int direct; - int format, sformat, n; + snd_pcm_format_t format, sformat; + int n; struct snd_mask sformat_mask; struct snd_mask mask; @@ -868,11 +869,11 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) _snd_pcm_hw_param_min(sparams, SNDRV_PCM_HW_PARAM_PERIODS, 2, 0); snd_mask_none(&mask); if (atomic_read(&substream->mmap_count)) - snd_mask_set(&mask, SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); + snd_mask_set(&mask, (__force int)SNDRV_PCM_ACCESS_MMAP_INTERLEAVED); else { - snd_mask_set(&mask, SNDRV_PCM_ACCESS_RW_INTERLEAVED); + snd_mask_set(&mask, (__force int)SNDRV_PCM_ACCESS_RW_INTERLEAVED); if (!direct) - snd_mask_set(&mask, SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); + snd_mask_set(&mask, (__force int)SNDRV_PCM_ACCESS_RW_NONINTERLEAVED); } err = snd_pcm_hw_param_mask(substream, sparams, SNDRV_PCM_HW_PARAM_ACCESS, &mask); if (err < 0) { @@ -891,19 +892,22 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) else sformat = snd_pcm_plug_slave_format(format, &sformat_mask); - if (sformat < 0 || !snd_mask_test(&sformat_mask, sformat)) { - for (sformat = 0; sformat <= SNDRV_PCM_FORMAT_LAST; sformat++) { - if (snd_mask_test(&sformat_mask, sformat) && + if ((__force int)sformat < 0 || + !snd_mask_test(&sformat_mask, (__force int)sformat)) { + for (sformat = (__force snd_pcm_format_t)0; + (__force int)sformat <= (__force int)SNDRV_PCM_FORMAT_LAST; + sformat = (__force snd_pcm_format_t)((__force int)sformat + 1)) { + if (snd_mask_test(&sformat_mask, (__force int)sformat) && snd_pcm_oss_format_to(sformat) >= 0) break; } - if (sformat > SNDRV_PCM_FORMAT_LAST) { + if ((__force int)sformat > (__force int)SNDRV_PCM_FORMAT_LAST) { snd_printd("Cannot find a format!!!\n"); err = -EINVAL; goto failure; } } - err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, sformat, 0); + err = _snd_pcm_hw_param_set(sparams, SNDRV_PCM_HW_PARAM_FORMAT, (__force int)sformat, 0); if (err < 0) goto failure; @@ -912,9 +916,9 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) } else { _snd_pcm_hw_params_any(params); _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_ACCESS, - SNDRV_PCM_ACCESS_RW_INTERLEAVED, 0); + (__force int)SNDRV_PCM_ACCESS_RW_INTERLEAVED, 0); _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_FORMAT, - snd_pcm_oss_format_from(runtime->oss.format), 0); + (__force int)snd_pcm_oss_format_from(runtime->oss.format), 0); _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, 0); _snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_RATE, @@ -1185,10 +1189,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const if (in_kernel) { mm_segment_t fs; fs = snd_enter_user(); - ret = snd_pcm_lib_write(substream, (void __user *)ptr, frames); + ret = snd_pcm_lib_write(substream, (void __force __user *)ptr, frames); snd_leave_user(fs); } else { - ret = snd_pcm_lib_write(substream, (void __user *)ptr, frames); + ret = snd_pcm_lib_write(substream, (void __force __user *)ptr, frames); } if (ret != -EPIPE && ret != -ESTRPIPE) break; @@ -1230,10 +1234,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p if (in_kernel) { mm_segment_t fs; fs = snd_enter_user(); - ret = snd_pcm_lib_read(substream, (void __user *)ptr, frames); + ret = snd_pcm_lib_read(substream, (void __force __user *)ptr, frames); snd_leave_user(fs); } else { - ret = snd_pcm_lib_read(substream, (void __user *)ptr, frames); + ret = snd_pcm_lib_read(substream, (void __force __user *)ptr, frames); } if (ret == -EPIPE) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { @@ -1333,7 +1337,7 @@ static ssize_t snd_pcm_oss_write2(struct snd_pcm_substream *substream, const cha struct snd_pcm_plugin_channel *channels; size_t oss_frame_bytes = (runtime->oss.plugin_first->src_width * runtime->oss.plugin_first->src_format.channels) / 8; if (!in_kernel) { - if (copy_from_user(runtime->oss.buffer, (const char __user *)buf, bytes)) + if (copy_from_user(runtime->oss.buffer, (const char __force __user *)buf, bytes)) return -EFAULT; buf = runtime->oss.buffer; } @@ -1429,7 +1433,7 @@ static ssize_t snd_pcm_oss_read2(struct snd_pcm_substream *substream, char *buf, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t frames, frames1; #ifdef CONFIG_SND_PCM_OSS_PLUGINS - char __user *final_dst = (char __user *)buf; + char __user *final_dst = (char __force __user *)buf; if (runtime->oss.plugin_first) { struct snd_pcm_plugin_channel *channels; size_t oss_frame_bytes = (runtime->oss.plugin_last->dst_width * runtime->oss.plugin_last->dst_format.channels) / 8; @@ -1549,6 +1553,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) { struct snd_pcm_runtime *runtime; ssize_t result = 0; + snd_pcm_state_t state; long res; wait_queue_t wait; @@ -1570,9 +1575,9 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) result = 0; set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_lock_irq(substream); - res = runtime->status->state; + state = runtime->status->state; snd_pcm_stream_unlock_irq(substream); - if (res != SNDRV_PCM_STATE_RUNNING) { + if (state != SNDRV_PCM_STATE_RUNNING) { set_current_state(TASK_RUNNING); break; } @@ -1658,7 +1663,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) size1); size1 /= runtime->channels; /* frames */ fs = snd_enter_user(); - snd_pcm_lib_write(substream, (void __user *)runtime->oss.buffer, size1); + snd_pcm_lib_write(substream, (void __force __user *)runtime->oss.buffer, size1); snd_leave_user(fs); } } else if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) { diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 6751daa3bb5..71cc3ddf5c1 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -264,7 +264,7 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc return frames; } -static int snd_pcm_plug_formats(struct snd_mask *mask, int format) +static int snd_pcm_plug_formats(struct snd_mask *mask, snd_pcm_format_t format) { struct snd_mask formats = *mask; u64 linfmts = (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | @@ -276,16 +276,16 @@ static int snd_pcm_plug_formats(struct snd_mask *mask, int format) SNDRV_PCM_FMTBIT_U24_3BE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE); - snd_mask_set(&formats, SNDRV_PCM_FORMAT_MU_LAW); + snd_mask_set(&formats, (__force int)SNDRV_PCM_FORMAT_MU_LAW); if (formats.bits[0] & (u32)linfmts) formats.bits[0] |= (u32)linfmts; if (formats.bits[1] & (u32)(linfmts >> 32)) formats.bits[1] |= (u32)(linfmts >> 32); - return snd_mask_test(&formats, format); + return snd_mask_test(&formats, (__force int)format); } -static int preferred_formats[] = { +static snd_pcm_format_t preferred_formats[] = { SNDRV_PCM_FORMAT_S16_LE, SNDRV_PCM_FORMAT_S16_BE, SNDRV_PCM_FORMAT_U16_LE, @@ -306,24 +306,25 @@ static int preferred_formats[] = { SNDRV_PCM_FORMAT_U8 }; -int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) +snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, + struct snd_mask *format_mask) { int i; - if (snd_mask_test(format_mask, format)) + if (snd_mask_test(format_mask, (__force int)format)) return format; - if (! snd_pcm_plug_formats(format_mask, format)) - return -EINVAL; + if (!snd_pcm_plug_formats(format_mask, format)) + return (__force snd_pcm_format_t)-EINVAL; if (snd_pcm_format_linear(format)) { unsigned int width = snd_pcm_format_width(format); int unsignd = snd_pcm_format_unsigned(format) > 0; int big = snd_pcm_format_big_endian(format) > 0; unsigned int badness, best = -1; - int best_format = -1; + snd_pcm_format_t best_format = (__force snd_pcm_format_t)-1; for (i = 0; i < ARRAY_SIZE(preferred_formats); i++) { - int f = preferred_formats[i]; + snd_pcm_format_t f = preferred_formats[i]; unsigned int w; - if (!snd_mask_test(format_mask, f)) + if (!snd_mask_test(format_mask, (__force int)f)) continue; w = snd_pcm_format_width(f); if (w >= width) @@ -337,17 +338,20 @@ int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) best = badness; } } - return best_format >= 0 ? best_format : -EINVAL; + if ((__force int)best_format >= 0) + return best_format; + else + return (__force snd_pcm_format_t)-EINVAL; } else { switch (format) { case SNDRV_PCM_FORMAT_MU_LAW: for (i = 0; i < ARRAY_SIZE(preferred_formats); ++i) { - int format1 = preferred_formats[i]; - if (snd_mask_test(format_mask, format1)) + snd_pcm_format_t format1 = preferred_formats[i]; + if (snd_mask_test(format_mask, (__force int)format1)) return format1; } default: - return -EINVAL; + return (__force snd_pcm_format_t)-EINVAL; } } } @@ -359,7 +363,7 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, struct snd_pcm_plugin_format tmpformat; struct snd_pcm_plugin_format dstformat; struct snd_pcm_plugin_format srcformat; - int src_access, dst_access; + snd_pcm_access_t src_access, dst_access; struct snd_pcm_plugin *plugin = NULL; int err; int stream = snd_pcm_plug_stream(plug); @@ -641,7 +645,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str } int snd_pcm_area_silence(const struct snd_pcm_channel_area *dst_area, size_t dst_offset, - size_t samples, int format) + size_t samples, snd_pcm_format_t format) { /* FIXME: sub byte resolution and odd dst_offset */ unsigned char *dst; @@ -688,7 +692,7 @@ int snd_pcm_area_silence(const struct snd_pcm_channel_area *dst_area, size_t dst int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_offset, const struct snd_pcm_channel_area *dst_area, size_t dst_offset, - size_t samples, int format) + size_t samples, snd_pcm_format_t format) { /* FIXME: sub byte resolution and odd dst_offset */ char *src, *dst; diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index b9afab60371..a5035c2369a 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -46,7 +46,7 @@ struct snd_pcm_plugin_channel { }; struct snd_pcm_plugin_format { - int format; + snd_pcm_format_t format; unsigned int rate; unsigned int channels; }; @@ -58,7 +58,7 @@ struct snd_pcm_plugin { struct snd_pcm_plugin_format dst_format; /* destination format */ int src_width; /* sample width in bits */ int dst_width; /* sample width in bits */ - int access; + snd_pcm_access_t access; snd_pcm_sframes_t (*src_frames)(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t dst_frames); snd_pcm_sframes_t (*dst_frames)(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t src_frames); snd_pcm_sframes_t (*client_channels)(struct snd_pcm_plugin *plugin, @@ -125,7 +125,8 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_pcm_hw_params *slave_params); -int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask); +snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, + struct snd_mask *format_mask); int snd_pcm_plugin_append(struct snd_pcm_plugin *plugin); @@ -146,12 +147,12 @@ snd_pcm_sframes_t snd_pcm_plugin_client_channels(struct snd_pcm_plugin *plugin, int snd_pcm_area_silence(const struct snd_pcm_channel_area *dst_channel, size_t dst_offset, - size_t samples, int format); + size_t samples, snd_pcm_format_t format); int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, size_t src_offset, const struct snd_pcm_channel_area *dst_channel, size_t dst_offset, - size_t samples, int format); + size_t samples, snd_pcm_format_t format); void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index bbe25d8c450..c8171f5783c 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -25,7 +25,7 @@ #include "pcm_plugin.h" static void zero_areas(struct snd_pcm_plugin_channel *dvp, int ndsts, - snd_pcm_uframes_t frames, int format) + snd_pcm_uframes_t frames, snd_pcm_format_t format) { int dst = 0; for (; dst < ndsts; ++dst) { @@ -38,7 +38,7 @@ static void zero_areas(struct snd_pcm_plugin_channel *dvp, int ndsts, static inline void copy_area(const struct snd_pcm_plugin_channel *src_channel, struct snd_pcm_plugin_channel *dst_channel, - snd_pcm_uframes_t frames, int format) + snd_pcm_uframes_t frames, snd_pcm_format_t format) { dst_channel->enabled = 1; snd_pcm_area_copy(&src_channel->area, 0, &dst_channel->area, 0, frames, format); @@ -51,7 +51,7 @@ static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin, { int nsrcs, ndsts, dst; struct snd_pcm_plugin_channel *dvp; - int format; + snd_pcm_format_t format; if (snd_BUG_ON(!plugin || !src_channels || !dst_channels)) return -ENXIO; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6b4b1287b31..ee9abb2d900 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -211,9 +211,9 @@ static char *snd_pcm_format_names[] = { const char *snd_pcm_format_name(snd_pcm_format_t format) { - if (format >= ARRAY_SIZE(snd_pcm_format_names)) + if ((__force unsigned int)format >= ARRAY_SIZE(snd_pcm_format_names)) return "Unknown"; - return snd_pcm_format_names[format]; + return snd_pcm_format_names[(__force unsigned int)format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); @@ -269,12 +269,12 @@ static const char *snd_pcm_stream_name(int stream) static const char *snd_pcm_access_name(snd_pcm_access_t access) { - return snd_pcm_access_names[access]; + return snd_pcm_access_names[(__force int)access]; } static const char *snd_pcm_subformat_name(snd_pcm_subformat_t subformat) { - return snd_pcm_subformat_names[subformat]; + return snd_pcm_subformat_names[(__force int)subformat]; } static const char *snd_pcm_tstamp_mode_name(int mode) @@ -284,7 +284,7 @@ static const char *snd_pcm_tstamp_mode_name(int mode) static const char *snd_pcm_state_name(snd_pcm_state_t state) { - return snd_pcm_state_names[state]; + return snd_pcm_state_names[(__force int)state]; } #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 434af3c56d5..88f02e3866e 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -35,7 +35,10 @@ struct pcm_format_data { unsigned char silence[8]; /* silence data to fill */ }; -static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { +/* we do lots of calculations on snd_pcm_format_t; shut up sparse */ +#define INT __force int + +static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { [SNDRV_PCM_FORMAT_S8] = { .width = 8, .phys = 8, .le = -1, .signd = 1, .silence = {}, @@ -215,9 +218,9 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { int snd_pcm_format_signed(snd_pcm_format_t format) { int val; - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return -EINVAL; - if ((val = pcm_formats[format].signd) < 0) + if ((val = pcm_formats[(INT)format].signd) < 0) return -EINVAL; return val; } @@ -266,9 +269,9 @@ EXPORT_SYMBOL(snd_pcm_format_linear); int snd_pcm_format_little_endian(snd_pcm_format_t format) { int val; - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return -EINVAL; - if ((val = pcm_formats[format].le) < 0) + if ((val = pcm_formats[(INT)format].le) < 0) return -EINVAL; return val; } @@ -304,9 +307,9 @@ EXPORT_SYMBOL(snd_pcm_format_big_endian); int snd_pcm_format_width(snd_pcm_format_t format) { int val; - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return -EINVAL; - if ((val = pcm_formats[format].width) == 0) + if ((val = pcm_formats[(INT)format].width) == 0) return -EINVAL; return val; } @@ -323,9 +326,9 @@ EXPORT_SYMBOL(snd_pcm_format_width); int snd_pcm_format_physical_width(snd_pcm_format_t format) { int val; - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return -EINVAL; - if ((val = pcm_formats[format].phys) == 0) + if ((val = pcm_formats[(INT)format].phys) == 0) return -EINVAL; return val; } @@ -358,11 +361,11 @@ EXPORT_SYMBOL(snd_pcm_format_size); */ const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format) { - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return NULL; - if (! pcm_formats[format].phys) + if (! pcm_formats[(INT)format].phys) return NULL; - return pcm_formats[format].silence; + return pcm_formats[(INT)format].silence; } EXPORT_SYMBOL(snd_pcm_format_silence_64); @@ -382,16 +385,16 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int int width; unsigned char *dst, *pat; - if (format < 0 || format > SNDRV_PCM_FORMAT_LAST) + if ((INT)format < 0 || (INT)format > (INT)SNDRV_PCM_FORMAT_LAST) return -EINVAL; if (samples == 0) return 0; - width = pcm_formats[format].phys; /* physical width */ - pat = pcm_formats[format].silence; + width = pcm_formats[(INT)format].phys; /* physical width */ + pat = pcm_formats[(INT)format].silence; if (! width) return -EINVAL; /* signed or 1 byte data */ - if (pcm_formats[format].signd == 1 || width <= 8) { + if (pcm_formats[(INT)format].signd == 1 || width <= 8) { unsigned int bytes = samples * width / 8; memset(data, *pat, bytes); return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4be45e7be8a..ae42b6509ce 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -941,7 +941,7 @@ static struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. */ -int snd_pcm_stop(struct snd_pcm_substream *substream, int state) +int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) { return snd_pcm_action(&snd_pcm_action_stop, substream, state); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 99a485f1364..f2436d33fbf 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1052,7 +1052,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, } else { #ifdef CONFIG_COMPAT if (client->convert32 && snd_seq_ev_is_varusr(&event)) { - void *ptr = compat_ptr(event.data.raw32.d[1]); + void *ptr = (void __force *)compat_ptr(event.data.raw32.d[1]); event.data.ext.ptr = ptr; } #endif @@ -2407,7 +2407,7 @@ int snd_seq_kernel_client_ctl(int clientid, unsigned int cmd, void *arg) if (client == NULL) return -ENXIO; fs = snd_enter_user(); - result = snd_seq_do_ioctl(client, cmd, (void __user *)arg); + result = snd_seq_do_ioctl(client, cmd, (void __force __user *)arg); snd_leave_user(fs); return result; } @@ -2497,9 +2497,6 @@ static void snd_seq_info_dump_ports(struct snd_info_buffer *buffer, } -void snd_seq_info_pool(struct snd_info_buffer *buffer, - struct snd_seq_pool *pool, char *space); - /* exported to seq_info.c */ void snd_seq_info_clients_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 7fb55436287..7f50c143767 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -86,7 +86,7 @@ int snd_seq_dump_var_event(const struct snd_seq_event *event, if (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR) { char buf[32]; - char __user *curptr = (char __user *)event->data.ext.ptr; + char __user *curptr = (char __force __user *)event->data.ext.ptr; while (len > 0) { int size = sizeof(buf); if (len < size) @@ -157,7 +157,7 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char if (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR) { if (! in_kernel) return -EINVAL; - if (copy_from_user(buf, (void __user *)event->data.ext.ptr, len)) + if (copy_from_user(buf, (void __force __user *)event->data.ext.ptr, len)) return -EFAULT; return newlen; } @@ -343,7 +343,7 @@ int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event, tmp->event = src->event; src = src->next; } else if (is_usrptr) { - if (copy_from_user(&tmp->event, (char __user *)buf, size)) { + if (copy_from_user(&tmp->event, (char __force __user *)buf, size)) { err = -EFAULT; goto __error; } diff --git a/sound/core/seq/seq_memory.h b/sound/core/seq/seq_memory.h index 63e91431a29..4a2ec779b8a 100644 --- a/sound/core/seq/seq_memory.h +++ b/sound/core/seq/seq_memory.h @@ -24,6 +24,8 @@ #include #include +struct snd_info_buffer; + /* container for sequencer event (internal use) */ struct snd_seq_event_cell { struct snd_seq_event event; @@ -99,5 +101,7 @@ void snd_sequencer_memory_done(void); /* polling */ int snd_seq_pool_poll_wait(struct snd_seq_pool *pool, struct file *file, poll_table *wait); +void snd_seq_info_pool(struct snd_info_buffer *buffer, + struct snd_seq_pool *pool, char *space); #endif diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 3b9b550109c..a89948ae9e8 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -18,7 +18,7 @@ * a subset of information returned via ctl info callback */ struct link_ctl_info { - int type; /* value type */ + snd_ctl_elem_type_t type; /* value type */ int count; /* item count */ int min_val, max_val; /* min, max values */ }; -- cgit v1.2.3 From 03c2d87a2112a6548aa3f9635e76d3611c3df53c Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Thu, 17 Feb 2011 00:17:53 +0100 Subject: ALSA: ac97: replace open-coded, error-prone stuff with AC97 bit defines Use AC97 macros (sometimes already existing, or newly added) instead of error-prone repetition of open-coded values. Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 5 +++ sound/pci/ac97/ac97_codec.c | 83 ++++++++++++++++++++++++++------------------- 2 files changed, 54 insertions(+), 34 deletions(-) (limited to 'include/sound') diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index b602f475cdb..f1dcefe4532 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -96,6 +96,10 @@ #define AC97_FUNC_INFO 0x68 /* Function Information */ #define AC97_SENSE_INFO 0x6a /* Sense Details */ +/* volume controls */ +#define AC97_MUTE_MASK_MONO 0x8000 +#define AC97_MUTE_MASK_STEREO 0x8080 + /* slot allocation */ #define AC97_SLOT_TAG 0 #define AC97_SLOT_CMD_ADDR 1 @@ -138,6 +142,7 @@ #define AC97_BC_18BIT_ADC 0x0100 /* 18-bit ADC resolution */ #define AC97_BC_20BIT_ADC 0x0200 /* 20-bit ADC resolution */ #define AC97_BC_ADC_MASK 0x0300 +#define AC97_BC_3D_TECH_ID_MASK 0x7c00 /* Per-vendor ID of 3D enhancement */ /* general purpose */ #define AC97_GP_DRSS_MASK 0x0c00 /* double rate slot select */ diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index cb62d178b3e..2cc56b5c539 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -590,9 +590,9 @@ static int snd_ac97_put_volsw(struct snd_kcontrol *kcontrol, snd_ac97_page_restore(ac97, page_save); #ifdef CONFIG_SND_AC97_POWER_SAVE /* check analog mixer power-down */ - if ((val_mask & 0x8000) && + if ((val_mask & AC97_PD_EAPD) && (kcontrol->private_value & (1<<30))) { - if (val & 0x8000) + if (val & AC97_PD_EAPD) ac97->power_up &= ~(1 << (reg>>1)); else ac97->power_up |= 1 << (reg>>1); @@ -1035,20 +1035,20 @@ static int snd_ac97_dev_free(struct snd_device *device) static int snd_ac97_try_volume_mix(struct snd_ac97 * ac97, int reg) { - unsigned short val, mask = 0x8000; + unsigned short val, mask = AC97_MUTE_MASK_MONO; if (! snd_ac97_valid_reg(ac97, reg)) return 0; switch (reg) { case AC97_MASTER_TONE: - return ac97->caps & 0x04 ? 1 : 0; + return ac97->caps & AC97_BC_BASS_TREBLE ? 1 : 0; case AC97_HEADPHONE: - return ac97->caps & 0x10 ? 1 : 0; + return ac97->caps & AC97_BC_HEADPHONE ? 1 : 0; case AC97_REC_GAIN_MIC: - return ac97->caps & 0x01 ? 1 : 0; + return ac97->caps & AC97_BC_DEDICATED_MIC ? 1 : 0; case AC97_3D_CONTROL: - if (ac97->caps & 0x7c00) { + if (ac97->caps & AC97_BC_3D_TECH_ID_MASK) { val = snd_ac97_read(ac97, reg); /* if nonzero - fixed and we can't set it */ return val == 0; @@ -1104,7 +1104,10 @@ static void check_volume_resolution(struct snd_ac97 *ac97, int reg, unsigned cha *lo_max = *hi_max = 0; for (i = 0 ; i < ARRAY_SIZE(cbit); i++) { unsigned short val; - snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8)); + snd_ac97_write( + ac97, reg, + AC97_MUTE_MASK_STEREO | cbit[i] | (cbit[i] << 8) + ); /* Do the read twice due to buffers on some ac97 codecs. * e.g. The STAC9704 returns exactly what you wrote to the register * if you read it immediately. This causes the detect routine to fail. @@ -1139,14 +1142,14 @@ static void snd_ac97_change_volume_params2(struct snd_ac97 * ac97, int reg, int unsigned short val, val1; *max = 63; - val = 0x8080 | (0x20 << shift); + val = AC97_MUTE_MASK_STEREO | (0x20 << shift); snd_ac97_write(ac97, reg, val); val1 = snd_ac97_read(ac97, reg); if (val != val1) { *max = 31; } /* reset volume to zero */ - snd_ac97_write_cache(ac97, reg, 0x8080); + snd_ac97_write_cache(ac97, reg, AC97_MUTE_MASK_STEREO); } static inline int printable(unsigned int x) @@ -1183,16 +1186,16 @@ static int snd_ac97_cmute_new_stereo(struct snd_card *card, char *name, int reg, if (! snd_ac97_valid_reg(ac97, reg)) return 0; - mute_mask = 0x8000; + mute_mask = AC97_MUTE_MASK_MONO; val = snd_ac97_read(ac97, reg); if (check_stereo || (ac97->flags & AC97_STEREO_MUTES)) { /* check whether both mute bits work */ - val1 = val | 0x8080; + val1 = val | AC97_MUTE_MASK_STEREO; snd_ac97_write(ac97, reg, val1); if (val1 == snd_ac97_read(ac97, reg)) - mute_mask = 0x8080; + mute_mask = AC97_MUTE_MASK_STEREO; } - if (mute_mask == 0x8080) { + if (mute_mask == AC97_MUTE_MASK_STEREO) { struct snd_kcontrol_new tmp = AC97_DOUBLE(name, reg, 15, 7, 1, 1); if (check_amix) tmp.private_value |= (1 << 30); @@ -1268,9 +1271,11 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne err = snd_ctl_add(card, kctl); if (err < 0) return err; - snd_ac97_write_cache(ac97, reg, - (snd_ac97_read(ac97, reg) & 0x8080) | - lo_max | (hi_max << 8)); + snd_ac97_write_cache( + ac97, reg, + (snd_ac97_read(ac97, reg) & AC97_MUTE_MASK_STEREO) + | lo_max | (hi_max << 8) + ); return 0; } @@ -1332,7 +1337,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) return err; } - ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080; + ac97->regs[AC97_CENTER_LFE_MASTER] = AC97_MUTE_MASK_STEREO; /* build center controls */ if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) @@ -1410,8 +1415,12 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_pc_beep[idx], ac97))) < 0) return err; set_tlv_db_scale(kctl, db_scale_4bit); - snd_ac97_write_cache(ac97, AC97_PC_BEEP, - snd_ac97_read(ac97, AC97_PC_BEEP) | 0x801e); + snd_ac97_write_cache( + ac97, + AC97_PC_BEEP, + (snd_ac97_read(ac97, AC97_PC_BEEP) + | AC97_MUTE_MASK_MONO | 0x001e) + ); } /* build Phone controls */ @@ -1545,7 +1554,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build Simulated Stereo Enhancement control */ - if (ac97->caps & 0x0008) { + if (ac97->caps & AC97_BC_SIM_STEREO) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_STEREO_ENHANCEMENT], ac97))) < 0) return err; } @@ -1557,7 +1566,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } /* build Loudness control */ - if (ac97->caps & 0x0020) { + if (ac97->caps & AC97_BC_LOUDNESS) { if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_general[AC97_GENERAL_LOUDNESS], ac97))) < 0) return err; } @@ -2542,8 +2551,8 @@ void snd_ac97_resume(struct snd_ac97 *ac97) schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); /* FIXME: extra delay */ - ac97->bus->ops->write(ac97, AC97_MASTER, 0x8000); - if (snd_ac97_read(ac97, AC97_MASTER) != 0x8000) + ac97->bus->ops->write(ac97, AC97_MASTER, AC97_MUTE_MASK_MONO); + if (snd_ac97_read(ac97, AC97_MASTER) != AC97_MUTE_MASK_MONO) msleep(250); } else { end_time = jiffies + msecs_to_jiffies(100); @@ -2747,12 +2756,12 @@ static int master_mute_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem int rshift = (kcontrol->private_value >> 12) & 0x0f; unsigned short mask; if (shift != rshift) - mask = 0x8080; + mask = AC97_MUTE_MASK_STEREO; else - mask = 0x8000; - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, + mask = AC97_MUTE_MASK_MONO; + snd_ac97_update_bits(ac97, AC97_POWERDOWN, AC97_PD_EAPD, (ac97->regs[AC97_MASTER] & mask) == mask ? - 0x8000 : 0); + AC97_PD_EAPD : 0); } return err; } @@ -2765,7 +2774,10 @@ static int tune_mute_led(struct snd_ac97 *ac97) return -ENOENT; msw->put = master_mute_sw_put; snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */ + snd_ac97_update_bits( + ac97, AC97_POWERDOWN, + AC97_PD_EAPD, AC97_PD_EAPD /* mute LED on */ + ); ac97->scaps |= AC97_SCAP_EAPD_LED; return 0; } @@ -2780,12 +2792,12 @@ static int hp_master_mute_sw_put(struct snd_kcontrol *kcontrol, int rshift = (kcontrol->private_value >> 12) & 0x0f; unsigned short mask; if (shift != rshift) - mask = 0x8080; + mask = AC97_MUTE_MASK_STEREO; else - mask = 0x8000; - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, + mask = AC97_MUTE_MASK_MONO; + snd_ac97_update_bits(ac97, AC97_POWERDOWN, AC97_PD_EAPD, (ac97->regs[AC97_MASTER] & mask) == mask ? - 0x8000 : 0); + AC97_PD_EAPD : 0); } return err; } @@ -2801,7 +2813,10 @@ static int tune_hp_mute_led(struct snd_ac97 *ac97) snd_ac97_remove_ctl(ac97, "External Amplifier", NULL); snd_ac97_remove_ctl(ac97, "Headphone Playback", "Switch"); snd_ac97_remove_ctl(ac97, "Headphone Playback", "Volume"); - snd_ac97_update_bits(ac97, AC97_POWERDOWN, 0x8000, 0x8000); /* mute LED on */ + snd_ac97_update_bits( + ac97, AC97_POWERDOWN, + AC97_PD_EAPD, AC97_PD_EAPD /* mute LED on */ + ); return 0; } -- cgit v1.2.3 From 3674f19dabd15f9541079a588149a370d888f4e6 Mon Sep 17 00:00:00 2001 From: Benjamin Herrenschmidt Date: Fri, 25 Mar 2011 17:51:54 +1100 Subject: ALSA: vmalloc buffers should use normal mmap It's a big no-no to use pgprot_noncached() when mmap'ing such buffers into userspace since they are mapped cachable in kernel space. This can cause all sort of interesting things ranging from to garbled sound to lockups on various architectures. I've observed that usb-audio is broken on powerpc 4xx for example because of that. Also remove the now unused snd_pcm_lib_mmap_noncached(). It's an arch business to know when to use uncached mappings, there's already hacks for MIPS inside snd_pcm_default_mmap() and other archs are supposed to use dma_mmap_coherent(). (See my separate patch that adds dma_mmap_coherent() to powerpc) Signed-off-by: Benjamin Herrenschmidt CC: Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 +--- sound/core/pcm_native.c | 9 --------- 2 files changed, 1 insertion(+), 12 deletions(-) (limited to 'include/sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 430a9cc045e..e1bad113061 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1031,9 +1031,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif -int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, - struct vm_area_struct *area); -#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached +#define snd_pcm_lib_mmap_vmalloc NULL static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ae42b6509ce..fe5c8036beb 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3201,15 +3201,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ -/* mmap callback with pgprot_noncached */ -int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - return snd_pcm_default_mmap(substream, area); -} -EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); - /* * mmap DMA buffer */ -- cgit v1.2.3