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[alsa-devel] [PATCH 2/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

2/4 Add firmware cache #2: This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Short description: This patch implements a simple cache for the firmware files when CONFIG_PM is defined. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 @@ -43,12 +43,24 @@ static int get_firmware(const struct fir { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct fir static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudi pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- Giuliano.

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[alsa-devel] [PATCH 1/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

1/5 Add firmware cache #1: When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Short description: Changes the way the firmware is passed through functions. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2010-01-02 21:27:05.000000000 +0100 @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *c /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-02 21:27:05.000000000 +0100 @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOC static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct fir return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -175,15 +175,15 @@ static inline int check_asic_status(stru #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struc return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudi chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-02 21:27:05.000000000 +0100 @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudi static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chi /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *c /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *c return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -32,8 +32,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *c /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoau /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct e { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chi /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *c mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *c /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC n if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoau if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- Giuliano.

1 0

[alsa-devel] [PATCH 0/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

The following patch set adds suspend/resume support to all echoaudio drivers. It has been tested on an IndigoIOx and a Gina3G. Rawmidi device has been tested partially. PCM and controls interfaces have been tested more and I think it's reasonably safe. I also have a Gina24 and a Gina3G, but I didn't test suspend/resume with them because I couldn't manage to make PM test system work on my pmac. Echoaudio soundcards do not support power management and Dx PCI states. Anyway, the card can enter a "comatose" state by sending a command to the DSP that is supposed to turn parts of the hardware off and reduce power consumption. Doing that causes the loss of the card's internal state, thus it is impossible, upon resume, to make the card continue from the exact point where it was suspended. The driver restores all controls and PCM settings and the applications have to call prepare() to restart the substreams. Rawmidi is restarted automatically, but some data in the i/o buffers may be lost. -- Giuliano.

1 0

[alsa-devel] dmix and dsnoop
by Raymond Yau 14 Feb '10

14 Feb '10

why do dmix and dsnoop 's slave alway open subdevice 0 ? for those hardware mixing sound cards which has two or more playback subdevices or hda-intel which may have 3 recording subdevices when subdevice 0 is opened by another application , dmix / dsnoop always fail even when there are available subdevices aplay -v -Dplug:dmix test.wav ALSA lib pcm_dmix.c:1010:(snd_pcm_dmix_open) unable to open slave aplay: main:654: audio open error: Device or resource busy arecord -v -Dplug:dsnoop:1 -f cd -d20 tt.wav ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave arecord: main:654: audio open error: Device or resource busy

1 0

[alsa-devel] [PATCH 1/2] ASoC: Make pmdown_time a per-card setting
by Mark Brown 13 Feb '10

13 Feb '10

Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 4 +++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e6a6d10..d9d88dd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -521,6 +521,8 @@ struct snd_soc_card { int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); + int pmdown_time; + /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ca89c78..94b9cde 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -542,7 +542,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(pmdown_time)); + msecs_to_jiffies(card->pmdown_time)); } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(codec, @@ -1039,6 +1039,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) dev_dbg(card->dev, "All components present, instantiating\n"); /* Found everything, bring it up */ + card->pmdown_time = pmdown_time; + if (card->probe) { ret = card->probe(pdev); if (ret < 0) -- 1.6.6.1

3 4

Re: [alsa-devel] HP 6930p mute button/led out of sync; fixed by suspend/resume cycle
by drago01 12 Feb '10

12 Feb '10

On Sun, Jan 10, 2010 at 3:31 PM, drago01 <drago01(a)gmail.com> wrote: > Hi, > > I experience an odd issue with my HP6930p laptop. > > After boot pressing the mute button/led causes the audio to be muted > but the sate is not reflected in software (i.e the mixer does not show > the control as muted). > However pressing after a suspend/resume cycle it works as expected. > The hardware mute button/led and the software control are in sync > (pressing the hw button toggles the mixer to mute/unmute; muting the > software mixer results into the led turning red). > > This was with 2.6.31.9 and 2.6.32.1; also installing alsa-drivers from > source does not change this. > > It seems something is not initialized correctly during boot up but is > done on resume, any pointers on how to debug this? > (Resending to correct list)

2 5

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 12 Feb '10

12 Feb '10

The following changes since commit c6848bf566c7217a6090693ff5cc47091fa772f5: Paul Menzel (1): ASoC: Typo. s/Freecale/Freescale/ are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n Mark Brown (3): ASoC: Add WM2000 driver ASoC: Make pmdown_time a per-card setting ASoC: Make pmdown_time runtime configurable Thomas Weber (1): ASoC: Add ASoC support for Devkit8000 include/sound/soc.h | 2 + include/sound/wm2000.h | 26 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm2000.c | 888 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm2000.h | 79 ++++ sound/soc/omap/Kconfig | 8 +- sound/soc/omap/omap3beagle.c | 6 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 31 ++- 10 files changed, 1040 insertions(+), 7 deletions(-) create mode 100644 include/sound/wm2000.h create mode 100644 sound/soc/codecs/wm2000.c create mode 100644 sound/soc/codecs/wm2000.h

1 0

[alsa-devel] [PATCH] ASoC: fix compile breakage if CONFIG_SH_DMA_API=y && CONFIG_SND_SIU_MIGOR!=n
by Guennadi Liakhovetski 12 Feb '10

12 Feb '10

Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not break anyway. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/sh/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index a86696b..1066749 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_SH4_FSI config SND_SOC_SH4_SIU tristate depends on (SUPERH || ARCH_SHMOBILE) && HAVE_CLK + select DMA_ENGINE select DMADEVICES select SH_DMAE -- 1.6.2.4

2 1

[alsa-devel] Jack Detection ??
by Johnson, Charles F 12 Feb '10

12 Feb '10

I'm trying to understand the level of jack detection implemented in ALSA. I admit some amount of ignorance here - is this documented somewhere that I haven't been able to find yet ?? Thanks. Charles Johnson Ultra-Mobility Group UPSG Architecture Intel Corporation charles.f.johnson(a)intel.com

2 1

[alsa-devel] IEC958 / SPDIF raw capture
by Donald Tournier 12 Feb '10

12 Feb '10

Hi, We have an application in our institute that will require recording of a digital signal via an optical cable. We have to be able to guarantee perfect bitwise reproduction of the signal (the odd bit error is not a problem). We're thinking of using SPDIF and ALSA to capture the signal (obviously we'll have to make sure the signal is properly encoded to begin with). However, after some initial attempts, it's becoming clear that ALSA can't necessarily guarantee bitwise perfect capture. ALSA might change the format, or resample the signal after capture (although I've heard that can be turned off with "snd_pcm_hw_params_set_rate_resample (capture_handle, hw_params, 0)"). Also, the card that we've been playing with so far (SB Audigy2 ZS) apparently performs some low-pass digital filtering on the input stream anyway, and is therefore no use to us. So my question is: can this be done? Is there a way of setting up ALSA to produce an unaltered version of the SPDIF input stream? This will undoubtedly require a particular combination of hardware and driver. If so, which card/driver would be recommended? Ideally, we would like to capture the raw SPDIF subframes. Is that possible, and if so, how? Any help would be greatly appreciated! Thanks, Donald. -- Jacques-Donald Tournier (PhD) Brain Research Institute, Melbourne, Australia Tel: +61 (0)3 9496 4078

3 4

[alsa-devel] [PATCH] ALSA: hda - Add Macmini 3,1 support
by Luke Yelavich 12 Feb '10

12 Feb '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++++++ 2 files changed, 137 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f06f20..0bbdbb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -124,6 +124,7 @@ ALC882/883/885/888/889 asus-a7m ASUS A7M macpro MacPro support mb5 Macbook 5,1 + macmini3 Macmini 3,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection imac91 iMac 9,1 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c22497..b5a6ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -211,6 +211,7 @@ enum { ALC885_MACPRO, ALC885_MBP3, ALC885_MB5, + ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -6751,6 +6752,14 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { @@ -6999,6 +7008,35 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_macmini3_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_macmini3_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + /* * 2ch mode @@ -7243,6 +7281,21 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7617,6 +7670,53 @@ static struct hda_verb alc885_mb5_init_verbs[] = { { } }; +/* Macmini 3,1 */ +static struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7800,6 +7900,18 @@ static void alc885_mb5_automute(struct hda_codec *codec) } +static void alc885_macmini3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + static void alc885_mb5_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -7808,6 +7920,14 @@ static void alc885_mb5_unsol_event(struct hda_codec *codec, alc885_mb5_automute(codec); } +static void alc885_macmini3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -8974,6 +9094,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9157,6 +9278,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), {} /* terminator */ }; @@ -9238,6 +9360,20 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mb5_unsol_event, .init_hook = alc885_mb5_automute, }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_macmini3_unsol_event, + .init_hook = alc885_macmini3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, -- 1.6.6.1

1 0

[alsa-devel] [PATCH] ASoC: TWL4030: Add supply for audio serial interface control
by Peter Ujfalusi 11 Feb '10

11 Feb '10

The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This patch has been tested in OPTION 1 mode, where the Voice path is not in use. I have looked for dependencies in the Voice path regarding to the Audio interface enable bit, but I have not found any. If someone could test, if the Voice path also needs the AIF_EN bit set, than please send a patch, which adds this new supply to the DAPM rule for the Voice path. sound/soc/codecs/twl4030.c | 14 +++++++++++++- 1 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b32aeb3..277862e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x0c, /* REG_ATXR1PGA (0xB) */ 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ - 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ @@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "APLL Enable"}, {"ADC Virtual Right2", NULL, "APLL Enable"}, + {"ADC Virtual Left1", NULL, "AIF Enable"}, + {"ADC Virtual Right1", NULL, "AIF Enable"}, + {"ADC Virtual Left2", NULL, "AIF Enable"}, + {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- 1.6.6.1

4 5

[alsa-devel] [RESEND][PATCHv2 3/4] Add ASoC support for Devkit8000
by Thomas Weber 11 Feb '10

11 Feb '10

This patch expands the omap3beagle sound soc for the beagle board clone DevKit8000. Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/omap/Kconfig | 8 +++++--- sound/soc/omap/omap3beagle.c | 6 +++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..18ebdc7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -94,12 +94,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + depends on TWL4030_CORE && SND_OMAP_SOC + depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Beagleboard. + Say Y if you want to add support for SoC audio on the Beagleboard or + the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- 1.6.6.1

4 3

[alsa-devel] Report : targa-dig and channels on GT725
by gens.guillaume＠free.fr 11 Feb '10

11 Feb '10

Hello, I have to report a problem about my GT725 laptop. I use the model : targa-dig. Connect a headphone to my laptop doesn't mute integrated speakers. I can do it manually by muting channels (Center, LFE) to have the sound only on my headphones. I use the alsa-driver (alsa-driver-20100211) from Takashi Iwai. I have the same symptomes that the stock kernel (from Archlinux). PS: and the microphone doesn't seem to work. See my alsa-info on attached file.

1 0

[alsa-devel] [PATCHv3 3/4] Add ASoC support for DevKit8000
by Thomas Weber 11 Feb '10

11 Feb '10

This patch expands the omap3beagle sound soc for the beagle board clone Devkit8000. Change log V2 Expand omap3beagle sound soc for DevKit8000 support. V3 Separate Kconfig for Devkit8000. Signed-off-by: Thomas Weber <swirl(a)gmx.li> --- sound/soc/omap/Kconfig | 8 ++++++++ sound/soc/omap/Makefile | 1 + sound/soc/omap/omap3beagle.c | 6 +++--- 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..8520b75 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -101,6 +101,14 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_DEVKIT8000 + tristate "SoC Audio support for DevKit8000" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_DEVKIT8000 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the DevKit8000 + config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 19283e5..c8195c9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -29,5 +29,6 @@ obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_DEVKIT8000) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- 1.6.4.4

5 7

[alsa-devel] Sound distortion with Realtek ALC889A
by Damien Brossard 11 Feb '10

11 Feb '10

Hi. I recently bought a Gigabyte GA-MA78GM-S2H motherboard. This board uses the (in)famous (from what i've read for now) alc889a chip. While playing 5.1 content, I can hear noise or distortion coming from the center channel. I can not test my configuration with another OS, so I can't fully dismiss a HW problem. However, i made several tests with 5.1 and 2.0 speakers which points me to soundcard problem. I've made some tests with speaker-test (speaker-test -D surround51 -c 6) : 1) Each time a sound is played either on front left channel or rear left channel, a noise can be heared on the center channel (and subwoofer, this is confirmed pluging a stereo headphone on the center/lfe jack). No noise when a sound is played on right channels (front or rear). 2) In alsamixer, if I set the volume to 0 for the front or rear left channels, i can hear the noise on center/lfe when a sound is "played" on this channels 3) In alsamixer, if I mute the front or rear left channels, i can hear no more noise on center/lfe Complementary tests have been made with alsaplayer and using a simple mp3 file. In this configuration (stereo output), center and rear channels play the same sound as front channel. 1) If all channels are unmuted, I hear no noise. 2) If either the front channel or the rear channel are muted, I can hear the noise on center channel. I tried this with alsa-driver 1.0.20 and 1.0.21 with the same result, I recently went back to 1.0.20 due to kernel upgrade. Things that seem strange to me (maybe not so strange to alsa dev :)) : 1) alsamixer shows volumes in percentage, however when I set a channel volume to 50% and the master volume to 50%, I hope to find a resulting volume of 25%, not 0. 2) lspci reports my soundcard subsystem id as 1458:a022, but in /proc/asound/card0/codec#0 the subsystem id is 1458:a002 Googling around made me find few references (in alsa-devel list) to a similar problem : http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008942.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-August/009754.html The distortion / noise described here is exactly what i can hear with my system. Apparently the user solved the issue buying another soundcard. I'd like to be sure it is the only way to go. Attached is the file produced by alsa-info.sh Thanks in advance. Best regards Damien Brossard

2 3

[alsa-devel] Wierdness in alsa-driver configure script
by Eliot Blennerhassett 11 Feb '10

11 Feb '10

checking for kernel version... 2.6.28-17-generic checking for GCC version... ./configure: eval: line 5540: syntax error near unexpected token `)' ./configure: eval: line 5540: `my_compiler_version=4.3.3-5ubuntu4)' Kernel compiler: Used compiler: gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 gcc --version gives (yes, it is ugly!): gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 Copyright (C) 2008 Free Software Foundation, Inc. The following change works for this particular case, but perhaps there is a better answer? --- a/configure.in +++ b/configure.in @@ -363,7 +363,7 @@ AC_DEFUN([CHECK_COMPILER], [ ;; [[0-9]].[[0-9]]*) if test -z "$ac_compiler_version"; then - ac_compiler_version="$i" + ac_compiler_version=`echo $i | sed -e 's|[[^0-9]]*$[[0-9.]]\+$.*$|\1|'` fi ;; esac

2 1

[alsa-devel] unpadded USB MIDI commands
by Willem Jan Palenstijn 11 Feb '10

11 Feb '10

Hi, I recently investigated some trouble with sending MIDI via a USB MIDI cable (15ca:1806) to a Roland MT-32. In some cases, commands were being dropped. (This is what is reported on https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 . I'm the colleague mentioned by druon in note 0021737 there.) After we experimented a bit, it turned out that packets of 2 MIDI commands (8 bytes total) were being sent to the USB MIDI cable, and that if the first of these was a 2-byte MIDI command (such as a 0xC0 or 0xD0 command), the second command in the packet would get dropped. A workaround is to limit the packet size to 4 bytes, as is already happening on at least one other device (the ESI M4U referred to in usbmidi.c). I played with it some more this week, and it turns out that if the zero-padding from the USB MIDI commands is removed (violating the specs, as far as I understand), the second command does actually not get dropped. Example: When sending the two commands 0xD1 0x10 and 0x91 0x40 0x40 normally, it would produce: sending packet: [ 0d d1 10 00 09 91 40 40 ] When routing the MIDI output through the MIDI THRU port on the MT-32 to a MIDI port on an ES1371, the captured stream is: d1 10 and no note is played on the MT-32. (The 0x91 command is a note-on.) After hacking out the zero padding, it looks like this: sending packet: [ 0d d1 10 09 91 40 40 ] The re-captured stream is: d1 10 91 40 40, and a note is properly played. What do you think would be the right way to handle this problem? Don't zero-pad the outgoing packets for this specific device (and maybe a few others mentioned on bug #3752, if they end up suffering from the same bug)? Limit the outgoing packets to 4 bytes like with the ESI M4U for these devices? Add an option to the snd-usb-audio module to toggle this behaviour? Something else? Best regards, Willem Jan

2 1

[alsa-devel] Realtek ALC275
by Jason A. Donenfeld 11 Feb '10

11 Feb '10

Hello All, The new Vaio F series has a Realtek ALC 275, which is currently setup in patch_realtek.c to use the 269 configuration. Unfortunately this does not correctly enable the internal microphone, and the line-in is not tied to any mixer control. The attached file will enable the internal microphone, but will do so at the expense of the line-in. According to Takashi, "The problem is that the digital-mic of pin 0x12 has the exclusive connection to the selector 0x22, followed to ADC 0x11, while other analog input pins are connected to 0x23/0x24, to ADC 0x08/0x09. Thus, the driver requires the dynamic switching of ADCs, which isn't fully guaranteed to work. A similar situation is found for Conexant codecs, and a same style hack would be needed for the realtek codec." This bug is being tracked here: http://code.google.com/p/vaio-f11-linux/issues/detail?id=2 and here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4909 . I'm wondering if anyone has any advice on how to implement the necessary dynamic ADC switching. Thanks, Jason Donenfeld

1 0

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 10 Feb '10

10 Feb '10

The following changes since commit c50749de02f272be6e09b9016e13a17307d29066: Grazvydas Ignotas (1): ASoC: pandora: Add DAC regulator support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (1): ASoC: cs4270: enable regulators at probe time Mark Brown (1): ASoC: add phycore-ac97 sound support Paul Menzel (1): ASoC: Typo. s/Freecale/Freescale/ Peter Ujfalusi (1): ASoC: TWL4030: Add supply for audio serial interface control sound/soc/codecs/cs4270.c | 10 +++++ sound/soc/codecs/twl4030.c | 14 ++++++- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/Makefile | 2 + sound/soc/imx/phycore-ac97.c | 90 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 116 insertions(+), 2 deletions(-) create mode 100644 sound/soc/imx/phycore-ac97.c

2 1

[alsa-devel] [PATCH] ASoC: Typo. s/Freecale/Freescale/
by Paul Menzel 10 Feb '10

10 Feb '10

Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 5f006f0..c7d0fd9 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,5 +1,5 @@ config SND_IMX_SOC - tristate "SoC Audio for Freecale i.MX CPUs" + tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC && BROKEN select SND_PCM select FIQ -- 1.6.6.1

2 2

[alsa-devel] [PATCH] Add Phytec i.MX sound support for AC97
by Sascha Hauer 10 Feb '10

10 Feb '10

Hi Marc et all, The following patch adds support for Phytec i.MX boards in AC97 mode. With this patch only some board glue code is missing to enable sound for the first i.MX boards. I'll push the board code through my i.MX tree. Please consider including this patch for -next. Sascha

3 5

[alsa-devel] [PATCH] ALSA: ASoC: cs4270: enable regulators at probe time
by Daniel Mack 09 Feb '10

09 Feb '10

Enable the bulk regulators at probe time so we can safely disable them again when going to suspend without confusing the reference counter. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b54575..70c5edb 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -617,8 +617,17 @@ static int cs4270_probe(struct platform_device *pdev) if (ret < 0) goto error_free_pcms; + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret < 0) + goto error_free_regulators; + return 0; +error_free_regulators: + regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + error_free_pcms: snd_soc_free_pcms(socdev); @@ -638,6 +647,7 @@ static int cs4270_remove(struct platform_device *pdev) struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); + regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies); return 0; -- 1.6.3.3

4 5

[alsa-devel] [PATCH V2 3/4] Add ASoC support for Devkit8000
by Thomas Weber 09 Feb '10

09 Feb '10

This patch expands the omap3beagle sound soc for the beagle board clone DevKit8000. Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/omap/Kconfig | 8 +++++--- sound/soc/omap/omap3beagle.c | 6 +++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..1cdd541 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -94,12 +94,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + depends on TWL4030_CORE && SND_OMAP_SOC + depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Beagleboard. + Say Y if you want to add support for SoC audio on the Beagleboard or + the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- 1.6.4.4

2 1

[alsa-devel] At91 ac97 patch
by Sedji Gaouaou 09 Feb '10

09 Feb '10

Hi, I was looking at the latest kernel and I noticed that the patch reagrding the atmel ac97 for AT91 has not been applied: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/021865.ht… Should I resubmit a new one? Regards, Sedji

1 0

[alsa-devel] [PATCH] sound: use DEFINE_PCI_DEVICE_TABLE
by Alexey Dobriyan 09 Feb '10

09 Feb '10

Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to .devinit.rodata section, so they can be discarded in some cases, and make them const. Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> --- sound/oss/kahlua.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au8810.c | 2 +- sound/pci/au88x0/au8820.c | 2 +- sound/pci/au88x0/au8830.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs5530.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 67 files changed, 70 insertions(+), 70 deletions(-) --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -198,7 +198,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static struct pci_device_id id_tbl[] = { +static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_device_id snd_ad1889_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static struct pci_device_id snd_ali_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -145,7 +145,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static struct pci_device_id snd_als300_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -117,7 +117,7 @@ struct snd_card_als4000 { #endif }; -static struct pci_device_id snd_als4000_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -164,7 +164,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static struct pci_device_id snd_aw2_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -350,7 +350,7 @@ struct snd_azf3328 { #endif }; -static const struct pci_device_id snd_azf3328_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -795,7 +795,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static struct pci_device_id snd_bt87x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -964,7 +964,7 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1875,7 +1875,7 @@ static int snd_ca0106_resume(struct pci_dev *pci) #endif // PCI IDs -static struct pci_device_id snd_ca0106_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2796,7 +2796,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static struct pci_device_id snd_cmipci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3018,7 +3018,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static struct pci_device_id intel_82437vx[] = { + static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_cs4281_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static struct pci_device_id snd_cs46xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -58,7 +58,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static struct pci_device_id snd_cs5530_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static struct pci_device_id snd_cs5535audio_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -43,7 +43,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static struct pci_device_id ct_pci_dev_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -76,7 +76,7 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static struct pci_device_id snd_emu10k1_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1605,7 +1605,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_emu10k1x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -443,7 +443,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_audiopci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1938_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -551,7 +551,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1968_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -205,7 +205,7 @@ struct fm801 { #endif }; -static struct pci_device_id snd_fm801_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2664,7 +2664,7 @@ static void __devexit azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static struct pci_device_id azx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* ICH 6..10 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -106,7 +106,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static const struct pci_device_id snd_ice1712_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static const struct pci_device_id snd_vt1724_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -420,7 +420,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static struct pci_device_id snd_intel8x0_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static struct pci_device_id snd_intel8x0m_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal <gamal(a)alternex.com.br>"); -static struct pci_device_id snd_korg1212_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { { .vendor = 0x10b5, .device = 0x906d, --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -55,7 +55,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static struct pci_device_id snd_lx6464es_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -861,7 +861,7 @@ struct snd_m3 { /* * pci ids */ -static struct pci_device_id snd_m3_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -60,7 +60,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static struct pci_device_id snd_mixart_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static struct pci_device_id snd_nm256_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id hifier_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -72,7 +72,7 @@ enum { MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; -static struct pci_device_id oxygen_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -40,7 +40,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id xonar_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -94,7 +94,7 @@ enum { PCI_ID_LAST }; -static struct pci_device_id pcxhr_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -506,7 +506,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static struct pci_device_id snd_riptide_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -515,7 +515,7 @@ static struct pci_device_id snd_riptide_ids[] = { }; #ifdef SUPPORT_JOYSTICK -static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme32_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -231,7 +231,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme96_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -585,7 +585,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_hdsp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -512,7 +512,7 @@ static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = { }; -static struct pci_device_id snd_hdspm_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -314,7 +314,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_rme9652_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { { .vendor = 0x10ee, .device = 0x3fc4, --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); -static struct pci_device_id snd_sis7019_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static struct pci_device_id snd_sonic_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static struct pci_device_id snd_trident_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -401,7 +401,7 @@ struct via82xx { #endif }; -static struct pci_device_id snd_via82xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static struct pci_device_id snd_via82xx_modem_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static struct pci_device_id snd_vx222_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static struct pci_device_id snd_ymfpci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */

2 1

[alsa-devel] [PATCH] alsa-info.sh: Sort the list of loaded modules.
by Paul Menzel 08 Feb '10

08 Feb '10

Searching for loaded modules in this list is easier with the list sorted. Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- utils/alsa-info.sh | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index fe7c08d..5b4e045 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -127,7 +127,7 @@ withlsmod() { echo "!!All Loaded Modules" >> $FILE echo "!!------------------" >> $FILE echo "" >> $FILE - lsmod |awk {'print $1'} >> $FILE + lsmod |sort |awk {'print $1'} >> $FILE echo "" >> $FILE echo "" >> $FILE } -- 1.6.6.1

1 0

[alsa-devel] DMA interrupt not getting triggered(nailed the problem need solution)
by ANISH KUMAR 08 Feb '10

08 Feb '10

Hi All, This problem is due to ETIMEDOUT(timeout) happening in s3c_snd_lrsync function(called by trigger). As i understand this could be due I2S being broken. This basically happens because of improper initialization of CPU or CODEC. But if that is the case then it would not have produced sound initially also but that is not the case(initially it is ringing i.e. interrupt coming for a short time then after i am getting time out problem)?? After this sound is not coming this happens when it does the initialisation again using open, prepare,enqueue and so on. Please provide any pointers which can help me out. Thanks & High Regards, anish

3 2

Re: [alsa-devel] DMA interrupt not getting triggered(nailed the problem need solution)
by ANISH KUMAR 08 Feb '10

08 Feb '10

> If the initial hardware state happens to result in LRCLK being where the >lrsync() function is trying to get it then the driver will start up, but >the misclocking of the configuration will result in ALSA detecting an >error which will eventually result in a restart of the audio stream. From the lrsync function i know that it is just trying to check whether a particular bit is set or not.So according to my understanding this lrsync function will timeout if the clock itself is not coming. Please forgive for not understanding your reply. But i am not able to understand "where the lrsync() function is trying to get-FROM YOUR REPLY"?? And misclocking of the configuration??Please explain this term(googling didn't help much).

2 1

[alsa-devel] [PATCH -next] ASoC: pandora: Add DAC regulator support
by Grazvydas Ignotas 08 Feb '10

08 Feb '10

Pandora's external DAC is connected to VSIM TWL4030 supply, so let's start switching it too to save more power. Also DAC got it's own DAPM handler and the delay was removed (it gave on audible improvement). Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 37 +++++++++++++++++++++++++++++++------ 1 files changed, 31 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 68980c1..b978411 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -22,7 +22,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/gpio.h> -#include <linux/delay.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/pcm.h> @@ -40,6 +40,8 @@ #define PREFIX "ASoC omap3pandora: " +static struct regulator *omap3pandora_dac_reg; + static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, unsigned int fmt) { @@ -106,21 +108,31 @@ static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_CBS_CFS); } -static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, +static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) { + regulator_enable(omap3pandora_dac_reg); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); } else { - gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); - mdelay(1); gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + regulator_disable(omap3pandora_dac_reg); } return 0; } +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + else + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + + return 0; +} + /* * Audio paths on Pandora board: * @@ -130,7 +142,9 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, * |P| <--- TWL4030 <--------- Line In and MICs */ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { - SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM, + 0, 0, omap3pandora_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, 0, 0, NULL, 0, omap3pandora_hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -306,8 +320,18 @@ static int __init omap3pandora_soc_init(void) goto fail2; } + omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc"); + if (IS_ERR(omap3pandora_dac_reg)) { + pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", + dev_name(&omap3pandora_snd_device->dev), + PTR_ERR(omap3pandora_dac_reg)); + goto fail3; + } + return 0; +fail3: + platform_device_del(omap3pandora_snd_device); fail2: platform_device_put(omap3pandora_snd_device); fail1: @@ -320,6 +344,7 @@ module_init(omap3pandora_soc_init); static void __exit omap3pandora_soc_exit(void) { + regulator_put(omap3pandora_dac_reg); platform_device_unregister(omap3pandora_snd_device); gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); -- 1.6.3.3

4 8

Re: [alsa-devel] DMA interrupt not getting triggered(nailed the problem need solution)
by ANISH KUMAR 08 Feb '10

08 Feb '10

>If the initial hardware state happens to result in LRCLK being where the >lrsync() function is trying to get it then the driver will start up, but >the misclocking of the configuration will result in ALSA detecting an >error which will eventually result in a restart of the audio stream. Mark in my case it is not restarting the audio stream(though it is trying to call trigger again for restarting) because in s3c_snd_lrsync function again i am getting time out(ETIMEDOUT). iiscon = readl(s3c_i2s.regs + S3C_IISCON); if (iiscon & S3C_IISCON_LRI) break; This LRI register bit is not getting set properly causing timeout.How this situation is not coming always(4/10)?? And who should suppose to set it? If my machine driver had a problem then this should have come always?? Thanks for your reply.

2 1

[alsa-devel] ALSA Driver 1.0.22.1
by Darsen Lu 07 Feb '10

07 Feb '10

I just installed alsa-drivers 1.0.22.1 (the latest version to my understanding) on Ubuntu 9.10. After that I started noticing conflicts between different sound-playback applications. For example, if I watch a Flash video on Firefox, stop it, and then start playing an MP3 file on Totem (a Media Player), I don't hear any sound. Once I close Firefox the music from Totem starts coming out. This never happened before on my Linux system. I'd be very grateful if someone could give me advices! My /var/log/syslog shows the following whenever this happens: Feb 7 02:42:36 darsen-laptop pulseaudio[1789]: alsa-sink.c: Error opening PCM device front:0: Device or resource busy Feb 7 02:42:36 darsen-laptop pulseaudio[1789]: alsa-sink.c: Error opening PCM device front:0: Device or resource busy I also noticed the following message during bootup and was asked to report this issue to ALSA Developers. Feb 7 02:17:31 darsen-laptop pulseaudio[1789]: ratelimit.c: 179 events suppressed Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Feb 7 02:17:35 darsen-laptop pulseaudio[1789]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Feb 7 02:17:36 darsen-laptop pulseaudio[1789]: ratelimit.c: 182 events suppressed The good news is my microphone started working with the newly-installed ALSA drivers (It did not work with ALSA v1.0.20) My Linux kernel version is 2.6.31-19.general. I installed it using the Ubuntu Package Manager. On the other hand the ALSA driver was compiled from source code. ALSA was configured with the following options sudo ./configure -with-kernel=/usr/src/linux-headers-`uname -r` --with-cards=hda-intel "lspci -v" shows the following 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) Subsystem: Dell Device 02cf Flags: bus master, fast devsel, latency 0, IRQ 31 Memory at f6afc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel I confirmed that the correct version of driver was loaded into the kernel by typing "cat /proc/asound/version": Advanced Linux Sound Architecture Driver Version 1.0.22.1. Compiled on Feb 7 2010 for kernel 2.6.31-19-generic (SMP). Thanks, Darsen

2 2

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 05 Feb '10

05 Feb '10

The following changes since commit cb67286d6629ecb5bfc44071d664cf1cbd01a350: Peter Ujfalusi (1): ASoC: TWL4030: Module unloading fix are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Grazvydas Ignotas (2): ASoC: pandora: Add APLL supply to fix audio output ASoC: pandora: Add DAC regulator support Mark Brown (1): Merge branch 'for-2.6.33' into for-2.6.34 sound/soc/omap/omap3pandora.c | 43 ++++++++++++++++++++++++++++++++++++---- 1 files changed, 38 insertions(+), 5 deletions(-)

2 1

[alsa-devel] ASoC update for 2.6.33
by Mark Brown 05 Feb '10

05 Feb '10

The following changes since commit 5bbd4953a4fb5d8d597b4a53b8da97eee320b634: Anuj Aggarwal (1): ASoC: AM3517: ASoC driver not getting compiled are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Grazvydas Ignotas (1): ASoC: pandora: Add APLL supply to fix audio output sound/soc/omap/omap3pandora.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: pandora: Add APLL supply to fix audio output
by Grazvydas Ignotas 05 Feb '10

05 Feb '10

Pandora's external DAC is using 256*Fs output from the TWL4030 codec, and TWL4030 needs to have APLL enabled for it's 256*Fs output to function. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- I noticed this a bit late because TWL4030 codec used to have APLL enable hardcoded until that was clean up by not-so-old patch. Would be nice to see this in next -rc, as without this audio won't work completely on pandora (again).. sound/soc/omap/omap3pandora.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 71b2c16..68980c1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { }; static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"PCM DAC", NULL, "APLL Enable"}, {"Headphone Amplifier", NULL, "PCM DAC"}, {"Line Out", NULL, "PCM DAC"}, {"Headphone Jack", NULL, "Headphone Amplifier"}, -- 1.6.3.3

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Re: [alsa-devel] [Q] playing / recording mono on a stereo-only hardware
by Guennadi Liakhovetski 05 Feb '10

05 Feb '10

(re-adding the list) On Tue, 2 Feb 2010, Michael Trimarchi wrote: > Hi, > > Guennadi Liakhovetski wrote: > > On Sat, 30 Jan 2010, Michael Trimarchi wrote: > > > > > > > Guennadi Liakhovetski wrote: > > > > > > > Hi > > > > > > > > I know this question belongs rather to alsa-users, and I did ask it > > > > there, > > > > but got no solution so far. Maybe someone here can help. > > > > > > > > The recently posted SuperH SIU driver only supports stereo playback and > > > > capture (channels_min = channels_max = 2). Is there a way to tell > > > > alsa-lib > > > > to duplicate the mono-channel during playback and to mix recorded > > > > channels > > > > (or use one of them), if mono recording is requested? > > > > > > > I'm not an expert but I think that you can use plughw support of alsa-lib > > > > > > pcm.record { > > > type plug > > > slave { > > > pcm record_hw > > > } > > > route_policy average > > > rate_converter "linear" > > > } > > > > > > I think that automatically change rate and channel for supporting mono > > > channel > > > too > > > > > > > Thanks for the snippet, but, I'm afraid, it didn't help either. > > > Sorry can you explain a little bit more why it doesn't help you? I certainly gladly would explain this, if only I knew... It just doesn't change anything. Are you using it anywhere? If yes, with which hardware configuration and software (alsa-lib) versions? Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] RFC: workaround for 'azx_get_response timeout, switching to polling mode...'
by Maxim Levitsky 05 Feb '10

05 Feb '10

Hi, Although I thought that polling mode is harmless, it sometimes isn't. I found out that after few dozens of hibernate cycles, the timeout would happen again kicking out first MSI, and then whole RRIB (single_cmd). However, I also found out that if I resend the command that caused the timeout, it would complete normally. Even futher, it is possible to poll for this command, and send next normally. It just works. So interrupts do work, but sometimes (very rarely) are missed. It might be even a hardware bug. The patch I send allows to delay switch to polling till we get 3 such timeouts in a row. I tested this approach for about 130 hibernate cycles. Best regards, Maxim Levitsky

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Re: [alsa-devel] DMA interrupt not getting triggered
by ANISH KUMAR 05 Feb '10

05 Feb '10

>Which board are you trying this on? S3C6410 >There may be some improvements with the DMA code in newer kernels, it would >be worth trying 2.6.33 and seeing if that has the same problem. There was >a problem with the circular DMA code failing to be fully merged which caused >problems including a resource exhaustion. Ben it would be very difficult to take the entire change.There are lot of efforts already gone into to make it stable. Can you please tell me which all changes exactly is required to get rid of this problem?As to what all the files that needs merging as i am not a kernel expert. or Can you give me some patch that would solve this problem?

1 0

[alsa-devel] DMA interrupt not getting triggered
by ANISH KUMAR 05 Feb '10

05 Feb '10

I am facing problem related to DMA interrupt not getting triggered some times(out of 5 times one time it is happening) causing application to wait for the buffers(hw_ptr = 0) and no sound because of this. I am using samsung soc with DMA controller PL080.I tried increasing periods_min to 4 but still no result. I am using 2.6.29 kernel version.

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[alsa-devel] DMA interrupt not getting triggered
by ANISH KUMAR 05 Feb '10

05 Feb '10

These are the logs in normal case: Entered s3c24xx_pcm_hw_free Entered s3c24xx_pcm_hw_free Entered s3c24xx_pcm_open Entered s3c24xx_pcm_hw_params, line 330 Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c6410_audio_buffdone Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_terpoin Entered s3c6410_audio_buffdone In abnormal case: Entered s3c24xx_pcm_hw_free Entered s3c24xx_pcm_hw_free Entered s3c24xx_pcm_close, Entered s3c24xx_pcm_open Entered s3c24xx_pcm_hw_param Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer

1 0

[alsa-devel] [PATCH] Allow mux to have up to 256 sources. Log warning and return index 0 rather than error if DSP returns invalid value. (amixer fails if error returned on get)
by linux＠audioscience.com 05 Feb '10

05 Feb '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- pci/asihpi/asihpi.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index 0c7b4fb..4f82e9f 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -2187,7 +2187,7 @@ static int snd_asihpi_mux_get(struct snd_kcontrol *kcontrol, HPI_HandleError(HPI_Multiplexer_GetSource(phSubSys, hControl, &wSourceType, &wSourceIndex)); /* Should cache this search result! */ - for (s = 0; s < 16; s++) { + for (s = 0; s < 256; s++) { if (HPI_Multiplexer_QuerySource(phSubSys, hControl, s, &wSrcNodeType, &wSrcNodeIndex)) break; @@ -2198,7 +2198,10 @@ static int snd_asihpi_mux_get(struct snd_kcontrol *kcontrol, return 0; } } - return -EINVAL; + snd_printd(KERN_WARNING "Control %x failed to match mux source %hu %hu\n", + hControl, wSourceType, wSourceIndex); + ucontrol->value.enumerated.item[0] = 0; + return 0; } static int snd_asihpi_mux_put(struct snd_kcontrol *kcontrol, -- 1.7.0.rc0.8.ge3f67d

2 1

[alsa-devel] DMA interrupt not getting triggered
by ANISH KUMAR 05 Feb '10

05 Feb '10

1 0

[alsa-devel] [PATCH 0/4] ASOC: Add AM3517 EVM support in ASOC
by Anuj Aggarwal 05 Feb '10

05 Feb '10

This patch series add support for OMAP3517 / AM3517 EVM in ASOC. It also enables the required drivers - I2C and McBSP, along with Alsa SoC subsystem, in the default configuration for the EVM. Anuj Aggarwal (4): Audio: Adding OMAP3517 / AM3517 EVM support in ASOC Audio: Modifying Kconfig/Makefile for AM3517 EVM Audio: Modifying board-evm file for audio codec Audio: Modifying the defconfig for OMAP3517 / AM3517 EVM arch/arm/configs/am3517_evm_defconfig | 78 ++++++++++++- arch/arm/mach-omap2/board-am3517evm.c | 21 ++++ sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/am3517evm.c | 202 +++++++++++++++++++++++++++++++++ 5 files changed, 309 insertions(+), 3 deletions(-) create mode 100644 sound/soc/omap/am3517evm.c

4 5

[alsa-devel] ioctl problem - help please!!
by Rene E. Henriquez G. 04 Feb '10

04 Feb '10

Good day to everyone. I'm developing a system in which I need to gather some entropy from /dev/random. In order to increase the entropy, I'm using a daemon called audio-entropyd from Folkert van Heusden, but it has some problems that maybe you guys can help me with this. Does anybody know what might cause this: 2925 open("/dev/snd/pcmC0D0c", O_RDWR|O_NONBLOCK) = 5 2925 ioctl(5, 0x80184151, 0x7ffff3351890) = -1 EIO (Input/output error) to give EIO? The daemon just stops with the KILL signal. Thanks for reading this, I really need your assistance for this please. Regards, René.

2 3

[alsa-devel] [PATCH] ALSA: cosmetic: make hda intel interrupt name consistent with others
by Maxim Levitsky 04 Feb '10

04 Feb '10

This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11ce655..0adea96 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) { if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - "HDA Intel", chip)) { + "hda_intel", chip)) { printk(KERN_ERR "hda-intel: unable to grab IRQ %d, " "disabling device\n", chip->pci->irq); if (do_disconnect) -- 1.6.3.3

1 0

[alsa-devel] [PATCH] hda-analyzer: Avoid AttributeError when codec has no nodes
by Ozan Çağlayan 04 Feb '10

04 Feb '10

When traversing card/codec/nodes during treewidget creation, it's quite possible that codec.nodes raises an AttributeError if the current codec doesn't seem to have a nodes dictionary. Simply initialize it to an empty dictionary in the beginning of analyze(). Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- hda-analyzer/hda_codec.py | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/hda-analyzer/hda_codec.py b/hda-analyzer/hda_codec.py index cbea3f8..d587a28 100644 --- a/hda-analyzer/hda_codec.py +++ b/hda-analyzer/hda_codec.py @@ -818,6 +818,7 @@ class HDACodec: def analyze(self): self.afg = None self.mfg = None + self.nodes = {} self.vendor_id = self.param_read(AC_NODE_ROOT, PARAMS['VENDOR_ID']) self.subsystem_id = self.param_read(AC_NODE_ROOT, PARAMS['SUBSYSTEM_ID']) self.revision_id = self.param_read(AC_NODE_ROOT, PARAMS['REV_ID']) @@ -866,7 +867,6 @@ class HDACodec: nodes_count, nid = self.get_sub_nodes(self.afg) self.base_nid = nid - self.nodes = {} for i in range(nodes_count): self.nodes[nid] = HDANode(self, nid) nid += 1 -- 1.6.5.6

2 1

[alsa-devel] ALC268 - Driving mute led with GPIO
by Ozan Çağlayan 04 Feb '10

04 Feb '10

Hi, We have a laptop here (a local brand) which has a mute button with a LED. Currently the mute button works correctly but the led is non-functional. I've played through hda-analyzer and found out that it's enabled when you enable gpio3 and click out_dir. Clicking out_dir again and again switches it on and off. There's a mute led stuff in patch_sigmatel, sigmatel_spec has gpio_led and gpio_polarity information but alc_spec doesn't have. How should we hook/patch that information into the codec? I'll try to send the alsa-info output tomorrow. Thanks. Ozan

2 3

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 04 Feb '10

04 Feb '10

The following changes since commit 026384d614b827f368834860c9b623ce08439b7e: Daniel Mack (1): ASoC: fix PXA SSP port resume are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compilation breakage in sound/soc/sh/fsi.c Mark Brown (8): ASoC: Allow CODECs to ask soc-cache to suppress physical writes ASoC: Add a cache_sync bit to the CODEC structure ASoC: Convert WM8993 to use shared cache I/O code ASoC: Initial WM8993 regulator API hookup ASoC: Disable WM8993 regulators when turning bias off ASoC: Add support for BIAS_OFF when idle to WM8904 ASoC: Optimise WM8904 output stage power control ASoC: Add WM8912 DAC support Peter Ujfalusi (1): ASoC: TWL4030: Module unloading fix include/sound/soc.h | 2 + sound/soc/codecs/twl4030.c | 5 +- sound/soc/codecs/wm8904.c | 176 ++++++++++++++++++++++++------- sound/soc/codecs/wm8993.c | 247 ++++++++++++++++++++++--------------------- sound/soc/sh/fsi.c | 40 +++---- sound/soc/soc-cache.c | 36 ++++++- 6 files changed, 320 insertions(+), 186 deletions(-)

2 1

[alsa-devel] [PATCH] [ALSA] es1968: remove unused code and macros
by Jiri Kosina 04 Feb '10

04 Feb '10

Remove code that has been commented out since beginning of git history. Also remove macros that are not used throughout the driver code at all. Signed-off-by: Jiri Kosina <jkosina(a)suse.cz> --- sound/pci/es1968.c | 181 ---------------------------------------------------- 1 files changed, 0 insertions(+), 181 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a11f453..5c35075 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -173,19 +173,10 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define ESS_FMT_STEREO 0x01 #define ESS_FMT_16BIT 0x02 -#define DAC_RUNNING 1 -#define ADC_RUNNING 2 - /* Values for the ESM_LEGACY_AUDIO_CONTROL */ #define ESS_DISABLE_AUDIO 0x8000 #define ESS_ENABLE_SERIAL_IRQ 0x4000 -#define IO_ADRESS_ALIAS 0x0020 -#define MPU401_IRQ_ENABLE 0x0010 -#define MPU401_IO_ENABLE 0x0008 -#define GAME_IO_ENABLE 0x0004 -#define FM_IO_ENABLE 0x0002 -#define SB_IO_ENABLE 0x0001 /* Values for the ESM_CONFIG_A */ @@ -193,9 +184,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define PIC_SNOOP2 0x2000 #define SAFEGUARD 0x0800 #define DMA_CLEAR 0x0700 -#define DMA_DDMA 0x0000 -#define DMA_TDMA 0x0100 -#define DMA_PCPCI 0x0200 #define POST_WRITE 0x0080 #define PCI_TIMING 0x0040 #define SWAP_LR 0x0020 @@ -217,7 +205,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define RINGB_DIS_VALIDATION 0x0008 #define RINGB_EN_SPDIF 0x0010 #define RINGB_EN_2CODEC 0x0020 -#define RINGB_SING_BIT_DUAL 0x0040 /* ****Port Adresses**** */ @@ -231,7 +218,6 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define ESM_RING_BUS_DEST 0x34 #define ESM_RING_BUS_CONTR_A 0x36 #define ESM_RING_BUS_CONTR_B 0x38 -#define ESM_RING_BUS_SDO 0x3A /* WaveCache*/ #define WC_INDEX 0x10 @@ -239,15 +225,9 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define WC_CONTROL 0x14 /* ASSP*/ -#define ASSP_INDEX 0x80 -#define ASSP_MEMORY 0x82 -#define ASSP_DATA 0x84 #define ASSP_CONTROL_A 0xA2 #define ASSP_CONTROL_B 0xA4 #define ASSP_CONTROL_C 0xA6 -#define ASSP_HOSTW_INDEX 0xA8 -#define ASSP_HOSTW_DATA 0xAA -#define ASSP_HOSTW_IRQ 0xAC /* Midi */ #define ESM_MPU401_PORT 0x98 /* Others */ @@ -256,19 +236,11 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); #define IDR0_DATA_PORT 0x00 #define IDR1_CRAM_POINTER 0x01 #define IDR2_CRAM_DATA 0x02 -#define IDR3_WAVE_DATA 0x03 -#define IDR4_WAVE_PTR_LOW 0x04 -#define IDR5_WAVE_PTR_HI 0x05 -#define IDR6_TIMER_CTRL 0x06 #define IDR7_WAVE_ROMRAM 0x07 -#define WRITEABLE_MAP 0xEFFFFF -#define READABLE_MAP 0x64003F - /* PCI Register */ #define ESM_LEGACY_AUDIO_CONTROL 0x40 -#define ESM_ACPI_COMMAND 0x54 #define ESM_CONFIG_A 0x50 #define ESM_CONFIG_B 0x52 #define ESM_DDMA 0x60 @@ -280,149 +252,23 @@ MODULE_PARM_DESC(joystick, "Enable joystick."); /* Host IRQ Control Bits */ #define ESM_RESET_MAESTRO 0x8000 #define ESM_RESET_DIRECTSOUND 0x4000 -#define ESM_HIRQ_ClkRun 0x0100 #define ESM_HIRQ_HW_VOLUME 0x0040 -#define ESM_HIRQ_HARPO 0x0030 /* What's that? */ -#define ESM_HIRQ_ASSP 0x0010 #define ESM_HIRQ_DSIE 0x0004 #define ESM_HIRQ_MPU401 0x0002 -#define ESM_HIRQ_SB 0x0001 /* Host IRQ Status Bits */ #define ESM_MPU401_IRQ 0x02 -#define ESM_SB_IRQ 0x01 #define ESM_SOUND_IRQ 0x04 -#define ESM_ASSP_IRQ 0x10 #define ESM_HWVOL_IRQ 0x40 #define ESS_SYSCLK 50000000 #define ESM_BOB_FREQ 200 #define ESM_BOB_FREQ_MAX 800 -#define ESM_FREQ_ESM1 (49152000L / 1024L) /* default rate 48000 */ -#define ESM_FREQ_ESM2 (50000000L / 1024L) - /* APU Modes: reg 0x00, bit 4-7 */ -#define ESM_APU_MODE_SHIFT 4 -#define ESM_APU_MODE_MASK (0xf << 4) -#define ESM_APU_OFF 0x00 -#define ESM_APU_16BITLINEAR 0x01 /* 16-Bit Linear Sample Player */ -#define ESM_APU_16BITSTEREO 0x02 /* 16-Bit Stereo Sample Player */ #define ESM_APU_8BITLINEAR 0x03 /* 8-Bit Linear Sample Player */ -#define ESM_APU_8BITSTEREO 0x04 /* 8-Bit Stereo Sample Player */ -#define ESM_APU_8BITDIFF 0x05 /* 8-Bit Differential Sample Playrer */ -#define ESM_APU_DIGITALDELAY 0x06 /* Digital Delay Line */ -#define ESM_APU_DUALTAP 0x07 /* Dual Tap Reader */ -#define ESM_APU_CORRELATOR 0x08 /* Correlator */ #define ESM_APU_INPUTMIXER 0x09 /* Input Mixer */ -#define ESM_APU_WAVETABLE 0x0A /* Wave Table Mode */ #define ESM_APU_SRCONVERTOR 0x0B /* Sample Rate Convertor */ -#define ESM_APU_16BITPINGPONG 0x0C /* 16-Bit Ping-Pong Sample Player */ -#define ESM_APU_RESERVED1 0x0D /* Reserved 1 */ -#define ESM_APU_RESERVED2 0x0E /* Reserved 2 */ -#define ESM_APU_RESERVED3 0x0F /* Reserved 3 */ - -/* reg 0x00 */ -#define ESM_APU_FILTER_Q_SHIFT 0 -#define ESM_APU_FILTER_Q_MASK (3 << 0) -/* APU Filtey Q Control */ -#define ESM_APU_FILTER_LESSQ 0x00 -#define ESM_APU_FILTER_MOREQ 0x03 - -#define ESM_APU_FILTER_TYPE_SHIFT 2 -#define ESM_APU_FILTER_TYPE_MASK (3 << 2) -#define ESM_APU_ENV_TYPE_SHIFT 8 -#define ESM_APU_ENV_TYPE_MASK (3 << 8) -#define ESM_APU_ENV_STATE_SHIFT 10 -#define ESM_APU_ENV_STATE_MASK (3 << 10) -#define ESM_APU_END_CURVE (1 << 12) -#define ESM_APU_INT_ON_LOOP (1 << 13) -#define ESM_APU_DMA_ENABLE (1 << 14) - -/* reg 0x02 */ -#define ESM_APU_SUBMIX_GROUP_SHIRT 0 -#define ESM_APU_SUBMIX_GROUP_MASK (7 << 0) -#define ESM_APU_SUBMIX_MODE (1 << 3) -#define ESM_APU_6dB (1 << 4) -#define ESM_APU_DUAL_EFFECT (1 << 5) -#define ESM_APU_EFFECT_CHANNELS_SHIFT 6 -#define ESM_APU_EFFECT_CHANNELS_MASK (3 << 6) - -/* reg 0x03 */ -#define ESM_APU_STEP_SIZE_MASK 0x0fff - -/* reg 0x04 */ -#define ESM_APU_PHASE_SHIFT 0 -#define ESM_APU_PHASE_MASK (0xff << 0) -#define ESM_APU_WAVE64K_PAGE_SHIFT 8 /* most 8bit of wave start offset */ -#define ESM_APU_WAVE64K_PAGE_MASK (0xff << 8) - -/* reg 0x05 - wave start offset */ -/* reg 0x06 - wave end offset */ -/* reg 0x07 - wave loop length */ - -/* reg 0x08 */ -#define ESM_APU_EFFECT_GAIN_SHIFT 0 -#define ESM_APU_EFFECT_GAIN_MASK (0xff << 0) -#define ESM_APU_TREMOLO_DEPTH_SHIFT 8 -#define ESM_APU_TREMOLO_DEPTH_MASK (0xf << 8) -#define ESM_APU_TREMOLO_RATE_SHIFT 12 -#define ESM_APU_TREMOLO_RATE_MASK (0xf << 12) - -/* reg 0x09 */ -/* bit 0-7 amplitude dest? */ -#define ESM_APU_AMPLITUDE_NOW_SHIFT 8 -#define ESM_APU_AMPLITUDE_NOW_MASK (0xff << 8) - -/* reg 0x0a */ -#define ESM_APU_POLAR_PAN_SHIFT 0 -#define ESM_APU_POLAR_PAN_MASK (0x3f << 0) -/* Polar Pan Control */ -#define ESM_APU_PAN_CENTER_CIRCLE 0x00 -#define ESM_APU_PAN_MIDDLE_RADIUS 0x01 -#define ESM_APU_PAN_OUTSIDE_RADIUS 0x02 - -#define ESM_APU_FILTER_TUNING_SHIFT 8 -#define ESM_APU_FILTER_TUNING_MASK (0xff << 8) - -/* reg 0x0b */ -#define ESM_APU_DATA_SRC_A_SHIFT 0 -#define ESM_APU_DATA_SRC_A_MASK (0x7f << 0) -#define ESM_APU_INV_POL_A (1 << 7) -#define ESM_APU_DATA_SRC_B_SHIFT 8 -#define ESM_APU_DATA_SRC_B_MASK (0x7f << 8) -#define ESM_APU_INV_POL_B (1 << 15) - -#define ESM_APU_VIBRATO_RATE_SHIFT 0 -#define ESM_APU_VIBRATO_RATE_MASK (0xf << 0) -#define ESM_APU_VIBRATO_DEPTH_SHIFT 4 -#define ESM_APU_VIBRATO_DEPTH_MASK (0xf << 4) -#define ESM_APU_VIBRATO_PHASE_SHIFT 8 -#define ESM_APU_VIBRATO_PHASE_MASK (0xff << 8) - -/* reg 0x0c */ -#define ESM_APU_RADIUS_SELECT (1 << 6) - -/* APU Filter Control */ -#define ESM_APU_FILTER_2POLE_LOPASS 0x00 -#define ESM_APU_FILTER_2POLE_BANDPASS 0x01 -#define ESM_APU_FILTER_2POLE_HIPASS 0x02 -#define ESM_APU_FILTER_1POLE_LOPASS 0x03 -#define ESM_APU_FILTER_1POLE_HIPASS 0x04 -#define ESM_APU_FILTER_OFF 0x05 - -/* APU ATFP Type */ -#define ESM_APU_ATFP_AMPLITUDE 0x00 -#define ESM_APU_ATFP_TREMELO 0x01 -#define ESM_APU_ATFP_FILTER 0x02 -#define ESM_APU_ATFP_PAN 0x03 - -/* APU ATFP Flags */ -#define ESM_APU_ATFP_FLG_OFF 0x00 -#define ESM_APU_ATFP_FLG_WAIT 0x01 -#define ESM_APU_ATFP_FLG_DONE 0x02 -#define ESM_APU_ATFP_FLG_INPROCESS 0x03 - /* capture mixing buffer size */ #define ESM_MEM_ALIGN 0x1000 @@ -729,33 +575,6 @@ static u16 apu_get_register(struct es1968 *chip, u16 channel, u8 reg) return v; } -#if 0 /* ASSP is not supported */ - -static void assp_set_register(struct es1968 *chip, u32 reg, u32 value) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->reg_lock, flags); - outl(reg, chip->io_port + ASSP_INDEX); - outl(value, chip->io_port + ASSP_DATA); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static u32 assp_get_register(struct es1968 *chip, u32 reg) -{ - unsigned long flags; - u32 value; - - spin_lock_irqsave(&chip->reg_lock, flags); - outl(reg, chip->io_port + ASSP_INDEX); - value = inl(chip->io_port + ASSP_DATA); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - return value; -} - -#endif - static void wave_set_register(struct es1968 *chip, u16 reg, u16 value) { unsigned long flags; -- 1.6.4.2

2 1

[alsa-devel] [PATCH] ASoC: TWL4030: Module unloading fix
by Peter Ujfalusi 04 Feb '10

04 Feb '10

The module unloading path had several problems: - it freed up the private structure twice - it freed up the codec structure, which was allocated as part of the private structure - it did not freed up the reg_cache - it did not unregistered the dais and the codec Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e0106a5..b32aeb3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2152,8 +2152,6 @@ static int twl4030_soc_remove(struct platform_device *pdev) twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -2237,6 +2235,9 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) { struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + snd_soc_unregister_codec(&twl4030->codec); + kfree(twl4030->codec.reg_cache); kfree(twl4030); twl4030_codec = NULL; -- 1.6.6.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: Allow CODECs to ask soc-cache to suppress physical writes
by Mark Brown 03 Feb '10

03 Feb '10

Currently the soc-cache code will always write to the device, meaning that we need the device to be powered and active at pretty much all times the system is active. Allowing cache only writes lays some groundwork for future enhancements to allow devices to be put into a full off state when the audio subsystem is idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 1 + sound/soc/soc-cache.c | 26 ++++++++++++++++++++++++-- 2 files changed, 25 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4bbeb9f..4e8f14b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -423,6 +423,7 @@ struct snd_soc_codec { short reg_cache_step; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ /* dapm */ u32 pop_time; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 097e335..84b6916 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -38,6 +38,10 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -100,6 +104,10 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 2); if (ret == 2) return 0; @@ -153,6 +161,9 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < codec->reg_cache_size) cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -181,6 +192,9 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, if (!snd_soc_codec_volatile_register(codec, reg)) reg_cache[reg] = value; + if (codec->cache_only) + return 0; + if (codec->hw_write(codec->control_data, data, 3) == 3) return 0; else @@ -193,10 +207,14 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; if (reg >= codec->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -EINVAL; + return codec->hw_read(codec, reg); - else + } else { return cache[reg]; + } } #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) @@ -294,6 +312,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (reg < codec->reg_cache_size) cache[reg] = value; + + if (codec->cache_only) + return 0; + ret = codec->hw_write(codec->control_data, data, 3); if (ret == 3) return 0; -- 1.6.6

1 1

[alsa-devel] [PATCH] ASoC: fix compilation breakage in sound/soc/sh/fsi.c
by Guennadi Liakhovetski 03 Feb '10

03 Feb '10

ctrl_outl() has become void at some point, which breaks compilation of fsi.c. Make writing functions void, as their output is anyway not evaluated, and use __raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl respectively. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- => linux-next / 2.6.34 sound/soc/sh/fsi.c | 40 +++++++++++++++++----------------------- 1 files changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ebf3588..3c36d24 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -120,35 +120,35 @@ struct fsi_master { ************************************************************************/ -static int __fsi_reg_write(u32 reg, u32 data) +static void __fsi_reg_write(u32 reg, u32 data) { /* valid data area is 24bit */ data &= 0x00ffffff; - return ctrl_outl(data, reg); + __raw_writel(data, reg); } static u32 __fsi_reg_read(u32 reg) { - return ctrl_inl(reg); + return __raw_readl(reg); } -static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) { u32 val = __fsi_reg_read(reg); val &= ~mask; val |= data & mask; - return __fsi_reg_write(reg, val); + __fsi_reg_write(reg, val); } -static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_write((u32)(fsi->base + reg), data); + __fsi_reg_write((u32)(fsi->base + reg), data); } static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) @@ -159,28 +159,25 @@ static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) return __fsi_reg_read((u32)(fsi->base + reg)); } -static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) { if (reg > REG_END) - return -1; + return; - return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); + __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); } -static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) +static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_write((u32)(master->base + reg), data); + __fsi_reg_write((u32)(master->base + reg), data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) @@ -199,21 +196,18 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg) return ret; } -static int fsi_master_mask_set(struct fsi_master *master, +static void fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { - int ret; unsigned long flags; if ((reg < MREG_START) || (reg > MREG_END)) - return -1; + return; spin_lock_irqsave(&master->lock, flags); - ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + __fsi_reg_mask_set((u32)(master->base + reg), mask, data); spin_unlock_irqrestore(&master->lock, flags); - - return ret; } /************************************************************************ -- 1.6.2.4

3 2

[alsa-devel] Access to bytes from the raw audio stream? (AC'97)
by John Graham 03 Feb '10

03 Feb '10

Hi there, I'm new to ALSA, and working on a project where I need to find out where I can see/modify the raw bytes coming in from/going out to a device. The device is an AC'97 codec (a philips UCB1400, if it makes any difference...) on an embedded board. I tried looking on the web, and found the tutorial at http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/, but that seems to be more about device management and configuration. So, can anybody tell me what functions and files in the sound/ directory of the linux kernel I need to look at to see/modify the raw data stream coming in? Am I just missing something in the tutorial? I'd appreciate any suggestions very much. Thanks, John Graham

2 3

[alsa-devel] amixer scontents fails, amixer contents succeeds
by Eliot Blennerhassett 03 Feb '10

03 Feb '10

Greetings, Probably a bug in my driver (asihpi), but I don't understand how "amixer controls" can work while "amixer scontrols" fails. (with error code -EINVAL: "Mixer hw:1 load error: Invalid argument") Where should I start looking for why this happens? thanks -- Eliot

2 1

[alsa-devel] Zoom R16 (USB), will a quirk make it work
by Jani Frilander 03 Feb '10

03 Feb '10

Hi. I have a Zoom R16 which audio interface part doesn't work with linux. lsusb -v: Bus 001 Device 004: ID 1686:00dd ZOOM Corporation Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x1686 ZOOM Corporation idProduct 0x00dd bcdDevice 0.02 iManufacturer 1 ZOOM Corporation iProduct 2 R16 iSerial 3 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 259 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 480mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0b 24 01 00 01 35 00 03 01 02 03 ** UNRECOGNIZED: 0c 24 02 05 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 08 01 03 00 05 00 ** UNRECOGNIZED: 0c 24 02 09 01 02 00 08 00 00 00 00 ** UNRECOGNIZED: 09 24 03 0c 01 01 00 09 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 05 01 01 00 ** UNRECOGNIZED: 14 24 02 01 02 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x006c 1x 108 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 0c 01 01 00 ** UNRECOGNIZED: 14 24 02 01 08 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 41 00 ** UNRECOGNIZED: 06 24 02 01 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 09 24 03 01 03 01 02 01 00 ** UNRECOGNIZED: 09 24 03 02 04 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered I have added following lines to usbquirks.h: /* Zoom R16 */ { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = 0x1686, .idProduct = 0x00dd, .bInterfaceClass = USB_CLASS_AUDIO, }, But it doesn't work. If I try to play a file with mplayer: $ mplayer -msglevel ao=9 -ao alsa:device=hw=2 piisi.mp3 MPlayer UNKNOWN-4.4.1 (C) 2000-2009 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing piisi.mp3. Audio only file format detected. ========================================================================== Opening audio decoder: [mp3lib] MPEG layer-2, layer-3 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000->176400) Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3) ========================================================================== Trying preferred audio driver 'alsa', options 'device=hw=2' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.20 alsa-init: setup for 1/2 channel(s) alsa-init: using device hw:2 alsa-init: pcm opened in blocking mode [AO_ALSA] Format s16le is not supported by hardware, trying default. [AO_ALSA] Unable to set format: Invalid argument Failed to initialize audio driver 'alsa:device=hw=2' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video $ cat /proc/asound/cards 0 [Audigy ]: Audigy - SB Audigy 1 [SB0090] SB Audigy 1 [SB0090] (rev.3, serial:0x511102) at 0x9c00, irq 17 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe024000 irq 27 2 [R16 ]: USB-Audio - R16 ZOOM Corporation R16 at usb-0000:00:0b.1-6, high speed $ ls /dev/snd/ by-id controlC2 midiC0D0 midiC2D0 pcmC0D2c pcmC1D0p seq by-path hwC0D0 midiC0D1 pcmC0D0c pcmC0D2p pcmC1D1p timer controlC0 hwC0D2 midiC0D2 pcmC0D0p pcmC0D3p pcmC2D0c controlC1 hwC1D0 midiC0D3 pcmC0D1c pcmC1D0c pcmC2D0p What can i do to make it work? I have a wireshark log of a winxp session. Does it help a bit? http://sites.google.com/site/zoomr16linux/zoomwire.bz2 Thanks for any advice. Sincerely, Jani Frilander

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[alsa-devel] [PATCH 1/4 v4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 03 Feb '10

03 Feb '10

The WM8978 codec from Wolfson Microelectronics is very similar to wm8974, but is stereo and also has some differences in pin configuration and internal signal routing. This driver is based on wm8974 and takes the differences into account. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v3 -> v4: 1. Removed "Mute" from two controls 2. Removed left-over "freq" parameter description 3. Now also suitably selecting f_PLLOUT, if OPCLK is not used 4. Switching PLL off for configuration 5. Switching PLL off and on and putting codec to sleep and waking up in suspend() / resume() diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62ff26a..0aad72f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C + select SND_SOC_WM8978 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C @@ -230,6 +231,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8974 tristate +config SND_SOC_WM8978 + tristate + config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea98354..fbd290e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8974-objs := wm8974.o +snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o +obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c new file mode 100644 index 0000000..b86ef47 --- /dev/null +++ b/sound/soc/codecs/wm8978.c @@ -0,0 +1,1124 @@ +/* + * wm8978.c -- WM8978 ALSA SoC Audio Codec driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * Copyright (C) 2007 Carlos Munoz <carlos(a)kenati.com> + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * Based on wm8974 and wm8990 by Liam Girdwood <lrg(a)slimlogic.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "wm8978.h" + +static struct snd_soc_codec *wm8978_codec; + +/* wm8978 register cache. Note that register 0 is not included in the cache. */ +static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ + 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ + 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ + 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ + 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ + 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ + 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ + 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ + 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ + 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ + 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ + 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ + 0x0001, 0x0001, /* 0x38...0x3b */ +}; + +/* codec private data */ +struct wm8978_priv { + struct snd_soc_codec codec; + unsigned int f_pllout; + unsigned int f_mclk; + unsigned int f_256fs; + unsigned int f_opclk; + enum wm8978_sysclk_src sysclk; + u16 reg_cache[WM8978_CACHEREGNUM]; +}; + +static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"}; +static const char *wm8978_eqmode[] = {"Capture", "Playback"}; +static const char *wm8978_bw[] = {"Narrow", "Wide"}; +static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"}; +static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"}; +static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"}; +static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"}; +static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; +static const char *wm8978_alc3[] = {"ALC", "Limiter"}; +static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; + +static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); + +static const struct snd_kcontrol_new wm8978_snd_controls[] = { + + SOC_SINGLE("Digital Loopback Switch", + WM8978_COMPANDING_CONTROL, 0, 1, 0), + + SOC_ENUM("ADC Companding", adc_compand), + SOC_ENUM("DAC Companding", dac_compand), + + SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("PCM Volume", + WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0), + SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Volume", + WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_ENUM("Equaliser Function", eqmode), + SOC_ENUM("EQ1 Cut Off", eq1), + SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ2 Bandwith", eq2bw), + SOC_ENUM("EQ2 Cut Off", eq2), + SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ3 Bandwith", eq3bw), + SOC_ENUM("EQ3 Cut Off", eq3), + SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ4 Bandwith", eq4bw), + SOC_ENUM("EQ4 Cut Off", eq4), + SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv), + + SOC_ENUM("EQ5 Cut Off", eq5), + SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv), + + SOC_SINGLE("DAC Playback Limiter Switch", + WM8978_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Decay", + WM8978_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Playback Limiter Attack", + WM8978_DAC_LIMITER_1, 0, 15, 0), + + SOC_SINGLE("DAC Playback Limiter Threshold", + WM8978_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE("DAC Playback Limiter Boost", + WM8978_DAC_LIMITER_2, 0, 15, 0), + + SOC_ENUM("ALC Enable Switch", alc1), + SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), + + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), + + SOC_ENUM("ALC Capture Mode", alc3), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", + WM8978_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + + /* OUT1 - Headphones */ + SOC_DOUBLE_R("Headphone Playback ZC Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT2 - Speakers */ + SOC_DOUBLE_R("Speaker Playback ZC Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Speaker Playback Volume", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT3/4 - Line Output */ + SOC_DOUBLE_R("Line Playback Switch", + WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1), + + /* Mixer #3: Boost (Input) mixer */ + SOC_DOUBLE_R("PGA Boost (+20dB)", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 0, 7, 0, boost_tlv), + + /* Input PGA volume */ + SOC_DOUBLE_R_TLV("Input PGA Volume", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 0, 63, 0, inpga_tlv), + + /* Headphone */ + SOC_DOUBLE_R("Headphone Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1), + + /* Speaker */ + SOC_DOUBLE_R("Speaker Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), +}; + +/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ +static const struct snd_kcontrol_new wm8978_left_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8978_right_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0), +}; + +/* OUT3/OUT4 Mixer not implemented */ + +/* Mixer #2: Input PGA Mute */ +static const struct snd_kcontrol_new wm8978_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new wm8978_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 1, 0), + + /* Mixer #1: OUT1,2 */ + SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3, + 2, 0, wm8978_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3, + 3, 0, wm8978_right_out_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2, + 2, 0, wm8978_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2, + 3, 0, wm8978_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2, + 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3, + 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3, + 8, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Output mixer */ + {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, + {"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left Output Mixer", "Aux Playback Switch", "LAUX"}, + {"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, +}; + +static int wm8978_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); + + /* set up the WM8978 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* PLL divisors */ +struct wm8978_pll_div { + u32 k; + u8 n; + u8 div2; +}; + +#define FIXED_PLL_SIZE (1 << 24) + +static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 k_part; + unsigned int k, n_div, n_mod; + + n_div = target / source; + if (n_div < 6) { + source >>= 1; + pll_div->div2 = 1; + n_div = target / source; + } else { + pll_div->div2 = 0; + } + + if (n_div < 6 || n_div > 12) + dev_warn(wm8978_codec->dev, + "WM8978 N value exceeds recommended range! N = %u\n", + n_div); + + pll_div->n = n_div; + n_mod = target - source * n_div; + k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2; + + do_div(k_part, source); + + k = k_part & 0xFFFFFFFF; + + pll_div->k = k; +} +/* + * Calculate internal frequencies and dividers, according to Figure 40 + * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 + */ +static int wm8978_configure_pll(struct snd_soc_codec *codec) +{ + struct wm8978_priv *wm8978 = codec->private_data; + struct wm8978_pll_div pll_div; + unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, + f_256fs = wm8978->f_256fs; + unsigned int f2, opclk_div; + + if (!f_mclk) + return -EINVAL; + + if (f_opclk) { + /* + * The user needs OPCLK. Choose OPCLKDIV to put + * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. + * f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4. + */ + if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk) + return -EINVAL; + + if (4 * f_opclk < 3 * f_mclk) + /* Have to use OPCLKDIV */ + opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk; + else + opclk_div = 1; + + dev_dbg(codec->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div); + + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 0x30, + (opclk_div - 1) << 4); + + wm8978->f_pllout = f_opclk * opclk_div; + } else if (f_256fs) { + /* + * Not using OPCLK, choose R: + * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. + * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors(). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK + * must be 3.781MHz <= f_MCLK <= 32.768MHz + */ + if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) + return -EINVAL; + + /* + * MCLKDIV will be selected in .hw_params(), just choose a + * suitable f_PLLOUT + */ + if (4 * f_256fs < 3 * f_mclk) + /* Will have to use MCLKDIV */ + wm8978->f_pllout = wm8978->f_mclk * 3 / 4; + else + wm8978->f_pllout = f_256fs; + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + } else { + return -EINVAL; + } + + f2 = wm8978->f_pllout * 4; + + dev_dbg(codec->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__, + wm8978->f_mclk, wm8978->f_pllout); + + pll_factors(&pll_div, f2, wm8978->f_mclk); + + dev_dbg(codec->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n", + __func__, pll_div.n, pll_div.k, pll_div.div2); + + /* Turn PLL off for configuration... */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + + snd_soc_write(codec, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n); + snd_soc_write(codec, WM8978_PLL_K1, pll_div.k >> 18); + snd_soc_write(codec, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8978_PLL_K3, pll_div.k & 0x1ff); + + /* ...and on again */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + if (f_opclk) + /* Output PLL (OPCLK) to GPIO1 */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 4); + + return 0; +} + +/* + * Configure WM8978 clock dividers. + */ +static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + switch (div_id) { + case WM8978_OPCLKRATE: + wm8978->f_opclk = div; + + if (wm8978->f_mclk) + ret = wm8978_configure_pll(codec); + break; + case WM8978_MCLKDIV: + if (div & ~0xe0) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); + break; + case WM8978_ADCCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); + break; + case WM8978_DACCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); + break; + case WM8978_BCLKDIV: + if (div & ~0x1c) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x1c, div); + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "%s: ID %d, value %u\n", __func__, div_id, div); + + return ret; +} + +/* + * @freq: when .set_pll() us not used, freq is codec MCLK input frequency + */ +static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq); + + if (freq) { + wm8978->f_mclk = freq; + + /* Even if MCLK is used for system clock, might have to drive OPCLK */ + if (wm8978->f_opclk) + ret = wm8978_configure_pll(codec); + + /* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */ + + if (!ret) + wm8978->sysclk = clk_id; + } + + if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) { + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + wm8978->sysclk = WM8978_MCLK; + wm8978->f_pllout = 0; + wm8978->f_opclk = 0; + } + + return ret; +} + +/* + * Set ADC and Voice DAC format. + */ +static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + /* + * BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80, + * Data Format mask = 0x18: all will be calculated anew + */ + u16 iface = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x198; + u16 clk = snd_soc_read(codec, WM8978_CLOCKING); + + dev_dbg(codec->dev, "%s\n", __func__); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + clk &= ~1; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x18; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); + snd_soc_write(codec, WM8978_CLOCKING, clk); + + return 0; +} + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8978_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + /* Word length mask = 0x60 */ + u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; + /* Sampling rate mask = 0xe (for filters) */ + u16 add_ctl = snd_soc_read(codec, WM8978_ADDITIONAL_CONTROL) & ~0xe; + u16 clking = snd_soc_read(codec, WM8978_CLOCKING); + enum wm8978_sysclk_src current_clk_id = clking & 0x100 ? + WM8978_PLL : WM8978_MCLK; + unsigned int f_sel, diff, diff_best = INT_MAX; + int i, best = 0; + + if (!wm8978->f_mclk) + return -EINVAL; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_ctl |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_ctl |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_ctl |= 0x60; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case 8000: + add_ctl |= 0x5 << 1; + break; + case 11025: + add_ctl |= 0x4 << 1; + break; + case 16000: + add_ctl |= 0x3 << 1; + break; + case 22050: + add_ctl |= 0x2 << 1; + break; + case 32000: + add_ctl |= 0x1 << 1; + break; + case 44100: + case 48000: + break; + } + + /* Sampling rate is known now, can configure the MCLK divider */ + wm8978->f_256fs = params_rate(params) * 256; + + if (wm8978->sysclk == WM8978_MCLK) { + f_sel = wm8978->f_mclk; + } else { + if (!wm8978->f_pllout) { + int ret = wm8978_configure_pll(codec); + if (ret < 0) + return ret; + } + f_sel = wm8978->f_pllout; + } + + /* + * In some cases it is possible to reconfigure PLL to a higher frequency + * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or + * 512 * fs, so, we should be fine. + */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + + if (diff < diff_best) { + diff_best = diff; + best = i; + } + + if (!diff) + break; + } + + if (diff) + dev_warn(codec->dev, "Imprecise clock: %u%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best], + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); + + dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, + params_format(params), params_rate(params), best); + + /* MCLK divisor mask = 0xe0 */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface_ctl); + snd_soc_write(codec, WM8978_ADDITIONAL_CONTROL, add_ctl); + + if (wm8978->sysclk != current_clk_id) { + if (wm8978->sysclk == WM8978_PLL) + /* Run CODEC from PLL instead of MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, + 0x100, 0x100); + else + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + } + + return 0; +} + +static int wm8978_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int wm8978_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 1; /* VMID 75k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_STANDBY: + /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ + power1 |= 0xc; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, + power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_OFF: + /* Preserve PLL - OPCLK may be used by someone */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, ~0x20, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); + + codec->bias_level = level; + return 0; +} + +#define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8978_dai_ops = { + .hw_params = wm8978_hw_params, + .digital_mute = wm8978_mute, + .set_fmt = wm8978_set_dai_fmt, + .set_clkdiv = wm8978_set_dai_clkdiv, + .set_sysclk = wm8978_set_dai_sysclk, +}; + +/* Also supports 12kHz */ +struct snd_soc_dai wm8978_dai = { + .name = "WM8978 HiFi", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .ops = &wm8978_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8978_dai); + +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* Also switch PLL off */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); + /* Put to sleep */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); + + return 0; +} + +static int wm8978_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int i; + u16 *cache = codec->reg_cache; + + /* Wake up the codec */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { + if (i == WM8978_RESET) + continue; + if (cache[i] != wm8978_reg[i]) + snd_soc_write(codec, i, cache[i]); + } + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (wm8978->f_pllout) + /* Switch PLL on */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + return 0; +} + +static int wm8978_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8978_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8978_codec; + codec = wm8978_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8978_snd_controls, + ARRAY_SIZE(wm8978_snd_controls)); + wm8978_add_widgets(codec); + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8978_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8978 = { + .probe = wm8978_probe, + .remove = wm8978_remove, + .suspend = wm8978_suspend, + .resume = wm8978_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8978); + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + WM8978_LEFT_DAC_DIGITAL_VOLUME, + WM8978_RIGHT_DAC_DIGITAL_VOLUME, + WM8978_LEFT_ADC_DIGITAL_VOLUME, + WM8978_RIGHT_ADC_DIGITAL_VOLUME, + WM8978_LEFT_INP_PGA_CONTROL, + WM8978_RIGHT_INP_PGA_CONTROL, + WM8978_LOUT1_HP_CONTROL, + WM8978_ROUT1_HP_CONTROL, + WM8978_LOUT2_SPK_CONTROL, + WM8978_ROUT2_SPK_CONTROL, +}; + +static __devinit int wm8978_register(struct wm8978_priv *wm8978) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8978->codec; + + if (wm8978_codec) { + dev_err(codec->dev, "Another WM8978 is registered\n"); + return -EINVAL; + } + + /* + * Set default system clock to PLL, it is more precise, this is also the + * default hardware setting + */ + wm8978->sysclk = WM8978_PLL; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8978; + codec->name = "WM8978"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8978_set_bias_level; + codec->dai = &wm8978_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8978_CACHEREGNUM; + codec->reg_cache = &wm8978->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8978_reg, sizeof(wm8978_reg)); + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + + /* Reset the codec */ + ret = snd_soc_write(codec, WM8978_RESET, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8978_dai.dev = codec->dev; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8978_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8978_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8978); + return ret; +} + +static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) +{ + wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8978_dai); + snd_soc_unregister_codec(&wm8978->codec); + kfree(wm8978); + wm8978_codec = NULL; +} + +static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8978_priv *wm8978; + struct snd_soc_codec *codec; + + wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + if (wm8978 == NULL) + return -ENOMEM; + + codec = &wm8978->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8978); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8978_register(wm8978); +} + +static __devexit int wm8978_i2c_remove(struct i2c_client *client) +{ + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + wm8978_unregister(wm8978); + return 0; +} + +static const struct i2c_device_id wm8978_i2c_id[] = { + { "wm8978", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); + +static struct i2c_driver wm8978_i2c_driver = { + .driver = { + .name = "WM8978", + .owner = THIS_MODULE, + }, + .probe = wm8978_i2c_probe, + .remove = __devexit_p(wm8978_i2c_remove), + .id_table = wm8978_i2c_id, +}; + +static int __init wm8978_modinit(void) +{ + return i2c_add_driver(&wm8978_i2c_driver); +} +module_init(wm8978_modinit); + +static void __exit wm8978_exit(void) +{ + i2c_del_driver(&wm8978_i2c_driver); +} +module_exit(wm8978_exit); + +MODULE_DESCRIPTION("ASoC WM8978 codec driver"); +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h new file mode 100644 index 0000000..b58f0bf --- /dev/null +++ b/sound/soc/codecs/wm8978.h @@ -0,0 +1,89 @@ +/* + * wm8978.h -- codec driver for WM8978 + * + * Copyright 2009 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WM8978_H__ +#define __WM8978_H__ + +/* + * Register values. + */ +#define WM8978_RESET 0x00 +#define WM8978_POWER_MANAGEMENT_1 0x01 +#define WM8978_POWER_MANAGEMENT_2 0x02 +#define WM8978_POWER_MANAGEMENT_3 0x03 +#define WM8978_AUDIO_INTERFACE 0x04 +#define WM8978_COMPANDING_CONTROL 0x05 +#define WM8978_CLOCKING 0x06 +#define WM8978_ADDITIONAL_CONTROL 0x07 +#define WM8978_GPIO_CONTROL 0x08 +#define WM8978_JACK_DETECT_CONTROL_1 0x09 +#define WM8978_DAC_CONTROL 0x0A +#define WM8978_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8978_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8978_JACK_DETECT_CONTROL_2 0x0D +#define WM8978_ADC_CONTROL 0x0E +#define WM8978_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8978_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8978_EQ1 0x12 +#define WM8978_EQ2 0x13 +#define WM8978_EQ3 0x14 +#define WM8978_EQ4 0x15 +#define WM8978_EQ5 0x16 +#define WM8978_DAC_LIMITER_1 0x18 +#define WM8978_DAC_LIMITER_2 0x19 +#define WM8978_NOTCH_FILTER_1 0x1b +#define WM8978_NOTCH_FILTER_2 0x1c +#define WM8978_NOTCH_FILTER_3 0x1d +#define WM8978_NOTCH_FILTER_4 0x1e +#define WM8978_ALC_CONTROL_1 0x20 +#define WM8978_ALC_CONTROL_2 0x21 +#define WM8978_ALC_CONTROL_3 0x22 +#define WM8978_NOISE_GATE 0x23 +#define WM8978_PLL_N 0x24 +#define WM8978_PLL_K1 0x25 +#define WM8978_PLL_K2 0x26 +#define WM8978_PLL_K3 0x27 +#define WM8978_3D_CONTROL 0x29 +#define WM8978_BEEP_CONTROL 0x2b +#define WM8978_INPUT_CONTROL 0x2c +#define WM8978_LEFT_INP_PGA_CONTROL 0x2d +#define WM8978_RIGHT_INP_PGA_CONTROL 0x2e +#define WM8978_LEFT_ADC_BOOST_CONTROL 0x2f +#define WM8978_RIGHT_ADC_BOOST_CONTROL 0x30 +#define WM8978_OUTPUT_CONTROL 0x31 +#define WM8978_LEFT_MIXER_CONTROL 0x32 +#define WM8978_RIGHT_MIXER_CONTROL 0x33 +#define WM8978_LOUT1_HP_CONTROL 0x34 +#define WM8978_ROUT1_HP_CONTROL 0x35 +#define WM8978_LOUT2_SPK_CONTROL 0x36 +#define WM8978_ROUT2_SPK_CONTROL 0x37 +#define WM8978_OUT3_MIXER_CONTROL 0x38 +#define WM8978_OUT4_MIXER_CONTROL 0x39 + +#define WM8978_CACHEREGNUM 58 + +/* Clock divider Id's */ +enum wm8978_clk_id { + WM8978_OPCLKRATE, + WM8978_MCLKDIV, + WM8978_ADCCLK, + WM8978_DACCLK, + WM8978_BCLKDIV, +}; + +enum wm8978_sysclk_src { + WM8978_PLL, + WM8978_MCLK +}; + +extern struct snd_soc_dai wm8978_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8978; + +#endif /* __WM8978_H__ */

3 7

[alsa-devel] About HDMI audio driver
by Wei Ni 03 Feb '10

03 Feb '10

Hi, Takashi I'm a nvidia engineer. We plan to develop the new hdmi audio driver for the new chipset. It support the latest HD Audio Specification. I download the latest kernel source, it has patch_intelhdmi.c. And this file have many defines and routines for HDMI operations. If I develop hdmi driver, it also need to define the same functions in patch_nvhdmi.c, it seems not very well. Do you have plan to change these codes to common code? I think these codes are common for HDMI audio. BTW I have a another question. Because the standard ALSA channel mapping is different with the Windows setting. So we could use channel-slot assignment (verb 0x734) to fix. How can I know the audio is using ALSA setting or Windows setting? Does there have any mechanism to support this operation? Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

4 4

[alsa-devel] [PATCH] Make firmware vs driver major version mismatch an error.
by linux＠audioscience.com 02 Feb '10

02 Feb '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> I.e. incompatible firmware will fail driver load. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- pci/asihpi/hpidspcd.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/pci/asihpi/hpidspcd.c b/pci/asihpi/hpidspcd.c index 8c0f251..f58d5a0 100644 --- a/pci/asihpi/hpidspcd.c +++ b/pci/asihpi/hpidspcd.c @@ -103,6 +103,13 @@ short HpiDspCode_Open( goto error2; } + if (header.version / 10000 != HPI_VER_DECIMAL / 10000) { + HPI_DEBUG_LOG(ERROR, + "Firmware Major Version mismatch DSP image %d != Driver %d\n", + header.version, HPI_VER_DECIMAL); + goto error2; + } + if (header.version != HPI_VER_DECIMAL) { HPI_DEBUG_LOG(WARNING, "Version mismatch DSP image %d != Driver %d\n", -- 1.7.0.rc0.8.ge3f67d

2 1

Re: [alsa-devel] [PATCH] Sync with AudioScience current CVS at version 4.03.04
by Eliot Blennerhassett 02 Feb '10

02 Feb '10

Greetings, The just submitted patch requires matching firmware http://audioscience.com/internet/download/firmware/dspbins40304.zip As usual, my apologies for the large patch. Unfortunately it was not possible to split into unrelated subpatches without manually recreating the commit history from our CVS repository. > b/pci/asihpi/Makefile | 2 > b/pci/asihpi/asihpi.c | 297 +++---- > b/pci/asihpi/hpi.h | 1446 ++++++++++++++++++------------------ > b/pci/asihpi/hpi6000.c | 102 +- > b/pci/asihpi/hpi6205.c | 678 +++++++++-------- > b/pci/asihpi/hpi6205.h | 23 > b/pci/asihpi/hpi_internal.h | 537 +++++++++++-- > b/pci/asihpi/hpicmn.c | 68 - > b/pci/asihpi/hpidebug.c | 40 - > b/pci/asihpi/hpidebug.h | 44 - > b/pci/asihpi/hpidspcd.h | 37 > b/pci/asihpi/hpifunc.c | 1571 ++++++++++++++++++++++++++-------------- > b/pci/asihpi/hpimsginit.c | 92 +- > b/pci/asihpi/hpimsginit.h | 33 > b/pci/asihpi/hpimsgx.c | 187 ++-- > b/pci/asihpi/hpioctl.c | 202 +++-- > b/pci/asihpi/hpios.c | 147 +++ > b/pci/asihpi/hpios.h | 234 ++++- > b/pci/asihpi/hpipcida.h | 6 > pci/asihpi/hpios_linux_kernel.c | 116 -- > pci/asihpi/hpios_linux_kernel.h | 189 ---- > 21 files changed, 3596 insertions(+), 2455 deletions(-) regards Eliot Blennerhassett AudioScience Inc.

3 2

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 02 Feb '10

02 Feb '10

The following changes since commit 9e6e96a197a03752d39a63e4f83e0b707ccedad7: Mark Brown (1): ASoC: Add WM8994 CODEC driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (1): ASoC: fix PXA SSP port resume Joe Perches (1): ASoC: Fix continuation line formats Joonyoung Shim (1): ASoC: Fix BCLK calculation of WM8994 Mark Brown (1): ASoC: Fix WM8994 dependency sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8994.c | 7 ++----- sound/soc/pxa/pxa-ssp.c | 12 +++++++----- sound/soc/s3c24xx/s3c-pcm.c | 3 +-- 4 files changed, 11 insertions(+), 13 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: ASoC: fix PXA SSP port resume
by Daniel Mack 02 Feb '10

02 Feb '10

Invalidate the cached hardware format on resume for PXA SSP ports. Otherwise hw_params() will bail out early at the next stream start, leaving the registers in a bogus state. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3bd7712..ebde673 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -146,6 +146,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) { struct ssp_priv *priv = cpu_dai->private_data; + /* the cached format information is invalid now */ + priv->dai_fmt = (unsigned int) -1; + if (!cpu_dai->active) return 0; -- 1.6.3.3

3 7

[alsa-devel] [PATCH 00/10] Non-printk continuation line fixes
by Joe Perches 02 Feb '10

02 Feb '10

Some more fixes to formats with continuation lines Joe Perches (10): arch/ia64: Fix continuation line formats drivers/ata: Fix continuation line formats drivers/gpu/drm/i915/intel_bios.c: Fix continuation line formats drivers/hwmon/vt8231.c: Fix continuation line formats drivers/net/davinci_emac.c: Fix continuation line formats drivers/scsi/qla2xxx/qua_os.c: Fix continuation line formats drivers/spi/spi_s3c64xx.c: Fix continuation line formats drivers/staging: Fix continuation line formats drivers/video/via: Fix continuation line formats sound/soc/s3c24xx/s2c-pcm.c: Fix continuation line formats arch/ia64/kvm/mmio.c | 4 ++-- arch/ia64/kvm/vcpu.c | 4 ++-- drivers/ata/pata_at91.c | 4 ++-- drivers/gpu/drm/i915/intel_bios.c | 3 +-- drivers/hwmon/vt8231.c | 3 +-- drivers/net/davinci_emac.c | 3 +-- drivers/scsi/qla2xxx/qla_os.c | 4 ++-- drivers/spi/spi_s3c64xx.c | 11 +++++------ drivers/staging/rtl8192su/r8192S_phy.c | 16 ++++++++-------- drivers/staging/sep/sep_driver.c | 5 +++-- drivers/staging/slicoss/slicoss.c | 6 ++---- drivers/video/via/dvi.c | 8 ++++---- drivers/video/via/lcd.c | 12 ++++++------ sound/soc/s3c24xx/s3c-pcm.c | 3 +-- 14 files changed, 40 insertions(+), 46 deletions(-)

2 2

[alsa-devel] [PATCH] ASoC: Fix BCLK calculation of WM8994
by Joonyoung Shim 02 Feb '10

02 Feb '10

Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 7bit Return-Path: jy0922.shim(a)samsung.com X-OriginalArrivalTime: 02 Feb 2010 09:41:49.0527 (UTC) FILETIME=[F1591670:01CAA3EB] This fixes BCLK calculation and removes unnecessary check code. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/wm8994.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5dd4b29..29f3771 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3267,15 +3267,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, */ best = 0; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - if (bclk_divs[i] < 0) - continue; - cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - - bclk_rate * 10; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - bclk_rate; if (cur_val < 0) /* BCLK table is sorted */ break; best = i; } - bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + bclk_rate = wm8994->aifclk[id] * 10 / bclk_divs[best]; dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", bclk_divs[best], bclk_rate); bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; -- 1.6.3.3

2 2

[alsa-devel] [PATCH 00/10] treewide: Fix format strings that misuse continuations
by Joe Perches 02 Feb '10

02 Feb '10

Format strings that are continued with \ are frequently misused. Change them to use mostly single line formats, some longer than 80 chars. Fix a few miscellaneous typos at the same time. Joe Perches (10): arch/powerpc: Fix continuation line formats arch/blackfin: Fix continuation line formats drivers/ide: Fix continuation line formats drivers/serial/bfin_5xx.c: Fix continuation line formats drivers/scsi/arcmsr: Fix continuation line formats drivers/staging: Fix continuation line formats drivers/net/amd8111e.c: Fix continuation line formats fs/proc/array.c: Fix continuation line formats mm/slab.c: Fix continuation line formats sound/soc/blackfin: Fix continuation line formats arch/blackfin/mach-common/smp.c | 4 +- arch/powerpc/kernel/nvram_64.c | 6 +- arch/powerpc/platforms/pseries/hotplug-cpu.c | 8 ++-- arch/powerpc/platforms/pseries/smp.c | 4 +- drivers/ide/au1xxx-ide.c | 4 +- drivers/ide/pmac.c | 4 +- drivers/net/amd8111e.c | 3 +- drivers/scsi/arcmsr/arcmsr_hba.c | 49 +++++++++---------- drivers/serial/bfin_5xx.c | 6 +-- drivers/staging/dream/qdsp5/audio_mp3.c | 3 +- .../rtl8187se/ieee80211/ieee80211_softmac_wx.c | 3 +- drivers/staging/rtl8187se/r8180_core.c | 3 +- drivers/staging/sep/sep_driver.c | 3 +- fs/proc/array.c | 7 ++- mm/slab.c | 4 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 8 +-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 3 +- 18 files changed, 55 insertions(+), 70 deletions(-)

3 5

[alsa-devel] Support for floating point controls?
by Eliot Blennerhassett 02 Feb '10

02 Feb '10

Greetings, some of our products now have control paramters that are 32 bit floating point values. I wonder if a patch implementing float ctl elements would be accepted? Note that the floats are not manipulated in the kernel at all, the values are just passed through from the hardware to userspace (?) Adding something like: #define SNDRV_CTL_ELEM_TYPE_FLOAT ((__force snd_ctl_elem_type_t) 7) struct snd_ctl_elem_info { ... union { struct { float min; /* R: minimum value */ float max; /* R: maximum value */ float step; /* R: step (0 variable) */ } float; ... } struct snd_ctl_elem_value { union { union { float value[128]; } float; -- Eliot Blennerhassett AudioScience Inc

5 6

[alsa-devel] SoC on PXA27x with two codecs
by Rodolfo Giometti 02 Feb '10

02 Feb '10

Hello, on my custom board I have a PXA27x CPU and 2 codecs: one connected as primary codec and one connected as secondary codec. I modify the SoC support in order to allow defining two audio cards, one to manage the first codec and one to manage the second one. Here the SoC schema: static struct snd_soc_dai_link bt270p_dai[] = { { .name = "AC97 HiFi", .stream_name = "audio out", .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], .codec_dai = &cs4299_dai[CS4299_DAI_AC97_HIFI], .init = bt270_ac97_init, .ops = &bt270_ops, }, { .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], .codec_dai = &cs4299_dai[CS4299_DAI_AC97_AUX], .init = bt270_ac97_init, .ops = &bt270_ops, }, }; static struct snd_soc_dai_link bt270s_dai[] = { { .name = "AC97 HiFi", .stream_name = "speakers", .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], .codec_dai = &cs4299_dai[CS4299_DAI_AC97_HIFI], .init = bt270_ac97_init, .ops = &bt270_ops, }, }; static struct snd_soc_card bt270_asoc[] = { { .name = "BT270 primary", .platform = &pxa2xx_soc_platform, .dai_link = bt270p_dai, .num_links = ARRAY_SIZE(bt270p_dai), }, { .name = "BT270 secondary", .platform = &pxa2xx_soc_platform, .dai_link = bt270s_dai, .num_links = ARRAY_SIZE(bt270s_dai), }, }; static struct cs4299_data cs4299_data[] = { { .ac97_line = 0, }, { .ac97_line = 1, }, }; static struct snd_soc_device bt270_snd_devdata[2] = { { .card = &bt270_asoc[0], .codec_dev = &soc_codec_dev_cs4299, .codec_data = &cs4299_data[0], }, { .card = &bt270_asoc[1], .codec_dev = &soc_codec_dev_cs4299, .codec_data = &cs4299_data[1], }, }; static struct platform_device *bt270_snd_device[2]; I'm using the ac97_line parameters into codec's code in order to select the primary and the second codecs as follow: static int cs4299_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct cs4299_data *data = (struct cs4299_data *) socdev->codec_data; struct snd_soc_codec *codec; int ac97_line; int ret = 0; printk(KERN_INFO "CS4299 SoC Audio Codec %s\n", CS4299_VERSION); /* Check specific codec data or set default if not defined */ ac97_line = 0; if (data) ac97_line = data->ac97_line; socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) return -ENOMEM; codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(cs4299_reg, sizeof(cs4299_reg), GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; } codec->reg_cache_size = sizeof(cs4299_reg); codec->reg_cache_step = 2; codec->name = "CS4299"; codec->id = ac97_line; codec->owner = THIS_MODULE; codec->dai = cs4299_dai; codec->num_dai = ARRAY_SIZE(cs4299_dai); codec->write = ac97_write; codec->read = ac97_read; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, ac97_line); if (ret < 0) { printk(KERN_ERR "cs4299: failed to register AC97 codec\n"); goto codec_err; } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) goto pcm_err; ret = cs4299_reset(codec, 0); if (ret < 0) { printk(KERN_ERR "Failed to reset CS4299: AC97 link error\n"); goto reset_err; } snd_soc_add_controls(codec, cs4299_snd_ac97_controls, ARRAY_SIZE(cs4299_snd_ac97_controls)); cs4299_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4299: failed to register card\n"); goto reset_err; } return 0; Everything is working ok but the PCM play/record. In fact I can manage both codecs' registers by using: alsamixer -c 0 and alsamixer -c 1 but when I try to play an MP3 file I can hear the music only on the second codec no matter which audio card I select for playing. I tested the hardware by defining just one codec at time, and doing in this manner the music goes to the right codec. Any suggeston? :) Thanks in advance, Rodolfo -- GNU/Linux Solutions e-mail: giometti(a)enneenne.com Linux Device Driver giometti(a)linux.it Embedded Systems phone: +39 349 2432127 UNIX programming skype: rodolfo.giometti Freelance ICT Italia - Consulente ICT Italia - www.consulenti-ict.it

2 1

[PATCH] ASoC: Fix BCLK calculation of WM8994
by Joonyoung Shim 02 Feb '10

02 Feb '10

This fixes BCLK calculation and removes unnecessary check code. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/wm8994.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 5dd4b29..29f3771 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3267,15 +3267,12 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, */ best = 0; for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { - if (bclk_divs[i] < 0) - continue; - cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - - bclk_rate * 10; + cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - bclk_rate; if (cur_val < 0) /* BCLK table is sorted */ break; best = i; } - bclk_rate = wm8994->aifclk[id] / bclk_divs[best]; + bclk_rate = wm8994->aifclk[id] * 10 / bclk_divs[best]; dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", bclk_divs[best], bclk_rate); bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; -- 1.6.3.3

1 0

[alsa-devel] alsa-pulse plugin
by Raymond Yau 02 Feb '10

02 Feb '10

When try to abort aplay -D pulse any.wav in a terminal on Fedora 12 the following error appear on some occasion and the terminal hang aplay -v any.wav ,,, ^CAborted by signal Interrupt... Assertion 'pthread_mutex_destroy(&m->mutex) == 0' failed at pulsecore/mutex-posix.c:83 function pa_mutex_free(). Aborting Aborted by signal Aborted.... *** glibc detected *** aplay: double free corruption(top): 0x09e69708 **** *** glibc detected *** aplay corrupted double-linked list : 0x09e5dcb8 *** Is there any way to debug this issue ?

2 1

[alsa-devel] [PATCH] trivial: sound seq ioctl dbg: print hexadecimal value padded with 0s
by Thadeu Lima de Souza Cascardo 02 Feb '10

02 Feb '10

Instead of padding with blanks and printing "number=0x a", print "number=0x0a". Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo(a)holoscopio.com> --- sound/core/seq/seq_clientmgr.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8ca2be3..48eca9f 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2190,7 +2190,7 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (p->cmd == cmd) return p->func(client, arg); } - snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%2x)\n", + snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); return -ENOTTY; } -- 1.6.6.1

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 02 Feb '10

02 Feb '10

The following changes since commit 2718625fba1e07bf2ce8a752036658737c1f76a7: Mark Brown (1): ASoC: Set codec->dev for AC97 devices are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Anuj Aggarwal (2): ASoC: AIC23: Fixing writes to non-existing registers in resume function ASoC: AM3517: ASoC driver not getting compiled Guennadi Liakhovetski (4): ASoC: add support for the sh7722 Migo-R board ASoC: remove bogus SLEEP mode from wm8978 driver ASoC: clean up wm8974 and wm8978 clock divider handling ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used Jassi Brar (1): ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset Joe Perches (1): ASoC: Fix continuation line formats Mark Brown (4): Merge branch 'for-2.6.33' into for-2.6.34 ASoC: Improved wm_hubs headphone handling ASoC: Activate DCS correction for WM8993 ASoC: Add WM8994 CODEC driver sound/soc/blackfin/bf5xx-ac97-pcm.c | 8 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 3 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 3 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/wm8974.c | 12 +- sound/soc/codecs/wm8974.h | 12 +- sound/soc/codecs/wm8978.c | 139 +- sound/soc/codecs/wm8978.h | 3 - sound/soc/codecs/wm8993.c | 11 +- sound/soc/codecs/wm8994.c | 3870 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.h | 26 + sound/soc/codecs/wm_hubs.c | 142 ++- sound/soc/codecs/wm_hubs.h | 6 + sound/soc/omap/Makefile | 2 +- sound/soc/s3c24xx/smdk_wm9713.c | 23 +- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/migor.c | 218 ++ 20 files changed, 4362 insertions(+), 134 deletions(-) create mode 100644 sound/soc/codecs/wm8994.c create mode 100644 sound/soc/codecs/wm8994.h create mode 100644 sound/soc/sh/migor.c

2 1

[alsa-devel] [PATCH] alsa: Add support for timberdale I2S block
by Richard Röjfors 01 Feb '10

01 Feb '10

This driver adds support for the I2S block of the timberdale FPGA. The timberdale is a FPGA found on intel development boards for In-Vehicle Infotainment. The block has support for up to 8 I2S channels, can be clocked from either the FPGA or the device side. This driver introduces support for this block, by exposing each I2S channel as an ALSA PCM channel. Signed-off-by: Richard Röjfors <richard.rojfors(a)mocean-labs.com> --- diff --git a/include/sound/timbi2s.h b/include/sound/timbi2s.h new file mode 100644 index 0000000..4ab810c --- /dev/null +++ b/include/sound/timbi2s.h @@ -0,0 +1,32 @@ +/* + * timbi2s.h timberdale FPGA I2S platform data + * Copyright (c) 2009 Intel Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ +#ifndef __INCLUDE_SOUND_TIMBI2S_H +#define __INCLUDE_SOUND_TIMBI2S_H + +struct timbi2s_bus_data { + u8 rx; + u16 sample_rate; +}; + +struct timbi2s_platform_data { + const struct timbi2s_bus_data *busses; + int num_busses; + u32 main_clk; +}; + +#endif diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 84714a6..54ad4e7 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -182,4 +182,17 @@ config SND_AC97_POWER_SAVE_DEFAULT The default time-out value in seconds for AC97 automatic power-save mode. 0 means to disable the power-save mode. +config SND_TIMBERDALE_I2S + tristate "The timberdale FPGA I2S driver" + depends on MFD_TIMBERDALE && HAS_IOMEM + default y + help + Say Y here to enable driver for the I2S block found within the + Timberdale FPGA. + There is support for up to 8 I2S channels, in either transmitter + or receiver mode. + + To compile this driver as a module, choose M here: the module + will be called snd-timbi2s. + endif # SND_DRIVERS diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index d4a07f9..dea2eed 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -10,6 +10,7 @@ snd-portman2x4-objs := portman2x4.o snd-serial-u16550-objs := serial-u16550.o snd-virmidi-objs := virmidi.o snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o +snd-timbi2s-objs := timbi2s.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o @@ -19,5 +20,6 @@ obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o obj-$(CONFIG_SND_MTS64) += snd-mts64.o obj-$(CONFIG_SND_PORTMAN2X4) += snd-portman2x4.o obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403-ac97cr.o +obj-$(CONFIG_SND_TIMBERDALE_I2S) += snd-timbi2s.o obj-$(CONFIG_SND) += opl3/ opl4/ mpu401/ vx/ pcsp/ diff --git a/sound/drivers/timbi2s.c b/sound/drivers/timbi2s.c new file mode 100644 index 0000000..a9c34c2 --- /dev/null +++ b/sound/drivers/timbi2s.c @@ -0,0 +1,755 @@ +/* + * timbi2s.c timberdale FPGA I2S driver + * Copyright (c) 2009 Intel Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +/* Supports: + * Timberdale FPGA I2S + * + */ + +#include <linux/io.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/timbi2s.h> + +#define DRIVER_NAME "timb-i2s" + +#define MAX_BUSSES 8 + +#define TIMBI2S_REG_VER 0x00 +#define TIMBI2S_REG_UIR 0x04 + +#define TIMBI2S_BUS_PRESCALE 0x00 +#define TIMBI2S_BUS_ICLR 0x04 +#define TIMBI2S_BUS_IPR 0x08 +#define TIMBI2S_BUS_ISR 0x0c +#define TIMBI2S_BUS_IER 0x10 + + +#define TIMBI2S_IRQ_TX_FULL 0x01 +#define TIMBI2S_IRQ_TX_ALMOST_FULL 0x02 +#define TIMBI2S_IRQ_TX_ALMOST_EMPTY 0x04 +#define TIMBI2S_IRQ_TX_EMPTY 0x08 + +#define TIMBI2S_IRQ_RX_FULL 0x10 +#define TIMBI2S_IRQ_RX_ALMOST_FULL 0x20 +#define TIMBI2S_IRQ_RX_ALMOST_EMPTY 0x40 +#define TIMBI2S_IRQ_RX_NOT_EMPTY 0x80 + +#define TIMBI2S_BUS_ICOR 0x14 +#define TIMBI2S_ICOR_TX_ENABLE 0x00000001 +#define TIMBI2S_ICOR_RX_ENABLE 0x00000002 +#define TIMBI2S_ICOR_LFIFO_RST 0x00000004 +#define TIMBI2S_ICOR_RFIFO_RST 0x00000008 +#define TIMBI2S_ICOR_FIFO_RST (TIMBI2S_ICOR_LFIFO_RST | TIMBI2S_ICOR_RFIFO_RST) +#define TIMBI2S_ICOR_SOFT_RST 0x00000010 +#define TIMBI2S_ICOR_WORD_SEL_LEFT_SHIFT 8 +#define TIMBI2S_ICOR_WORD_SEL_LEFT_MASK (0xff << 8) +#define TIMBI2S_ICOR_WORD_SEL_RIGHT_SHIFT 16 +#define TIMBI2S_ICOR_WORD_SEL_RIGHT_MASK (0xff << 16) +#define TIMBI2S_ICOR_CLK_MASTER 0x10000000 +#define TIMBI2S_ICOR_RX_ID 0x20000000 +#define TIMBI2S_ICOR_TX_ID 0x40000000 +#define TIMBI2S_ICOR_WORD_SEL 0x80000000 +#define TIMBI2S_BUS_FIFO 0x18 + +#define TIMBI2S_BUS_REG_AREA_SIZE (TIMBI2S_BUS_FIFO - \ + TIMBI2S_BUS_PRESCALE + 4) +#define TIMBI2S_FIRST_BUS_AREA_OFS 0x08 + +struct timbi2s_bus { + u32 flags; + u32 prescale; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_pcm_substream *substream; + unsigned buf_pos; + spinlock_t lock; /* mutual exclusion */ + u16 sample_rate; +}; + +#define BUS_RX 0x200 +#define BUS_MASTER 0x100 +#define BUS_INDEX_MASK 0xff +#define BUS_INDEX(b) ((b)->flags & BUS_INDEX_MASK) +#define BUS_IS_MASTER(b) ((b)->flags & BUS_MASTER) +#define BUS_IS_RX(b) ((b)->flags & BUS_RX) + +#define SET_BUS_INDEX(b, id) ((b)->flags = ((b)->flags & ~BUS_INDEX_MASK) | id) +#define SET_BUS_MASTER(b) ((b)->flags |= BUS_MASTER) +#define SET_BUS_RX(b) ((b)->flags |= BUS_RX) + +#define TIMBI2S_BUS_OFFSET(bus) (TIMBI2S_FIRST_BUS_AREA_OFS + \ + TIMBI2S_BUS_REG_AREA_SIZE * BUS_INDEX(bus)) + +struct timbi2s { + void __iomem *membase; + int irq; + struct tasklet_struct tasklet; + u32 main_clk; + unsigned num_busses; + struct timbi2s_bus busses[0]; +}; + +#define BITS_PER_CHANNEL 16 +#define NUM_CHANNELS 2 + +#define SAMPLE_SIZE ((NUM_CHANNELS * BITS_PER_CHANNEL) / 8) +#define NUM_PERIODS 32 +#define NUM_SAMPLES 256 + +static struct snd_pcm_hardware timbi2s_rx_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, /* only stereo */ + .channels_max = 2, + .buffer_bytes_max = NUM_PERIODS * SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_min = SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_max = SAMPLE_SIZE * NUM_SAMPLES, + .periods_min = NUM_PERIODS, + .periods_max = NUM_PERIODS, +}; + +static struct snd_pcm_hardware timbi2s_tx_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, /* only stereo */ + .channels_max = 2, + .buffer_bytes_max = NUM_PERIODS * SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_min = SAMPLE_SIZE * NUM_SAMPLES, + .period_bytes_max = SAMPLE_SIZE * NUM_SAMPLES, + .periods_min = NUM_PERIODS, + .periods_max = NUM_PERIODS, +}; + +static inline void timbi2s_bus_write(struct timbi2s_bus *bus, u32 val, u32 reg) +{ + struct timbi2s *i2s = snd_pcm_chip(bus->card); + + iowrite32(val, i2s->membase + TIMBI2S_BUS_OFFSET(bus) + reg); +} + +static inline u32 timbi2s_bus_read(struct timbi2s_bus *bus, u32 reg) +{ + struct timbi2s *i2s = snd_pcm_chip(bus->card); + + return ioread32(i2s->membase + TIMBI2S_BUS_OFFSET(bus) + reg); +} + +static u32 timbi2s_calc_prescale(u32 main_clk, u32 sample_rate) +{ + u32 halfbit_rate = sample_rate * BITS_PER_CHANNEL * NUM_CHANNELS * 2; + return main_clk / halfbit_rate; +} + +static int timbi2s_open(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct snd_pcm_runtime *runtime = substream->runtime; + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, substream, + BUS_INDEX(bus)); + + if (BUS_IS_RX(bus)) { + runtime->hw = timbi2s_rx_hw; + if (bus->sample_rate == 8000) { + runtime->hw.rates = SNDRV_PCM_RATE_8000; + runtime->hw.rate_min = 8000; + runtime->hw.rate_max = 8000; + } + } else + runtime->hw = timbi2s_tx_hw; + + bus->substream = substream; + + return 0; +} + +static int timbi2s_close(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, substream, + BUS_INDEX(bus)); + + bus->substream = NULL; + + return 0; +} + +static int timbi2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct timbi2s *i2s = snd_pcm_chip(card); + int err; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d\n", __func__, + substream, BUS_INDEX(bus)); + + bus->prescale = timbi2s_calc_prescale(i2s->main_clk, + params_rate(hw_params)); + + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + dev_dbg(snd_card_get_device_link(card), + "%s: Rate: %d, format: %d\n", __func__, params_rate(hw_params), + params_format(hw_params)); + + return 0; +} + +static int timbi2s_hw_free(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p\n", __func__, substream); + + spin_lock_irqsave(&bus->lock, flags); + /* disable interrupts */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + + /* disable TX and RX */ + timbi2s_bus_write(bus, TIMBI2S_ICOR_FIFO_RST | TIMBI2S_ICOR_SOFT_RST, + TIMBI2S_BUS_ICOR); + + return snd_pcm_lib_free_pages(substream); +} + +static int timbi2s_prepare(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + struct snd_pcm_runtime *runtime = substream->runtime; + u32 data; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, buffer: %d, period: %d\n", + __func__, substream, + BUS_INDEX(bus), (int)snd_pcm_lib_buffer_bytes(substream), + (int)snd_pcm_lib_period_bytes(substream)); + + if (runtime->dma_addr & 3 || runtime->buffer_size & 3) { + dev_err(snd_card_get_device_link(card), + "%s: Only word aligned data allowed\n", __func__); + return -EINVAL; + } + + if (runtime->channels != NUM_CHANNELS) { + dev_err(snd_card_get_device_link(card), + "%s: Number of channels unsupported %d\n", __func__, + runtime->channels); + return -EINVAL; + } + + /* reset */ + timbi2s_bus_write(bus, TIMBI2S_ICOR_FIFO_RST | TIMBI2S_ICOR_SOFT_RST, + TIMBI2S_BUS_ICOR); + + /* only masters have prescaling, don't write if not needed */ + if (BUS_IS_MASTER(bus)) + timbi2s_bus_write(bus, bus->prescale, TIMBI2S_BUS_PRESCALE); + + /* write word select */ + data = ((BITS_PER_CHANNEL << TIMBI2S_ICOR_WORD_SEL_LEFT_SHIFT) & + TIMBI2S_ICOR_WORD_SEL_LEFT_MASK) | + ((BITS_PER_CHANNEL << TIMBI2S_ICOR_WORD_SEL_RIGHT_SHIFT) & + TIMBI2S_ICOR_WORD_SEL_RIGHT_MASK); + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + + bus->buf_pos = 0; + + return 0; +} + +static int +timbi2s_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + u32 data; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, cmd: %d\n", __func__, + substream, BUS_INDEX(bus), cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_START command\n", __func__); + + /* start */ + data = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + data |= TIMBI2S_ICOR_TX_ENABLE; + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + + /* enable interrupts */ + timbi2s_bus_write(bus, TIMBI2S_IRQ_TX_ALMOST_EMPTY, + TIMBI2S_BUS_IER); + dev_dbg(snd_card_get_device_link(card), + "%s: ISR: %x, ICOR: %x\n", __func__, + timbi2s_bus_read(bus, TIMBI2S_BUS_ISR), + timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR)); + break; + case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_STOP command\n", __func__); + + spin_lock_irqsave(&bus->lock, flags); + /* disable interrupts */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + + /* reset */ + data = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + data &= ~TIMBI2S_ICOR_TX_ENABLE; + + timbi2s_bus_write(bus, data, TIMBI2S_BUS_ICOR); + break; + default: + dev_dbg(snd_card_get_device_link(card), + "%s: Got unsupported command\n", __func__); + + return -EINVAL; + } + + return 0; +} + +static int +timbi2s_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + unsigned long flags; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p, bus: %d, cmd: %d\n", __func__, + substream, BUS_INDEX(bus), cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_START command\n", __func__); + + timbi2s_bus_write(bus, TIMBI2S_ICOR_RX_ENABLE | + TIMBI2S_ICOR_FIFO_RST, TIMBI2S_BUS_ICOR); + + timbi2s_bus_write(bus, TIMBI2S_IRQ_RX_ALMOST_FULL, + TIMBI2S_BUS_IER); + break; + case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(snd_card_get_device_link(card), + "%s: Got TRIGGER_STOP command\n", __func__); + /* disable interrupts */ + spin_lock_irqsave(&bus->lock, flags); + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_IER); + spin_unlock_irqrestore(&bus->lock, flags); + /* Stop RX */ + timbi2s_bus_write(bus, 0, TIMBI2S_BUS_ICOR); + break; + default: + dev_dbg(snd_card_get_device_link(card), + "%s: Got unsupported command\n", __func__); + + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t +timbi2s_pointer(struct snd_pcm_substream *substream) +{ + struct timbi2s_bus *bus = snd_pcm_substream_chip(substream); + struct snd_card *card = bus->card; + snd_pcm_uframes_t ret; + + dev_dbg(snd_card_get_device_link(card), + "%s: Entry, substream: %p\n", __func__, substream); + + ret = bytes_to_frames(substream->runtime, bus->buf_pos); + if (ret >= substream->runtime->buffer_size) + ret -= substream->runtime->buffer_size; + + return ret; +} + +static struct snd_pcm_ops timbi2s_playback_ops = { + .open = timbi2s_open, + .close = timbi2s_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = timbi2s_hw_params, + .hw_free = timbi2s_hw_free, + .prepare = timbi2s_prepare, + .trigger = timbi2s_playback_trigger, + .pointer = timbi2s_pointer, +}; + +static struct snd_pcm_ops timbi2s_capture_ops = { + .open = timbi2s_open, + .close = timbi2s_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = timbi2s_hw_params, + .hw_free = timbi2s_hw_free, + .prepare = timbi2s_prepare, + .trigger = timbi2s_capture_trigger, + .pointer = timbi2s_pointer, +}; + +static void timbi2s_irq_process_rx(struct timbi2s_bus *bus) +{ + struct snd_pcm_runtime *runtime = bus->substream->runtime; + u32 buffer_size = snd_pcm_lib_buffer_bytes(bus->substream); + u32 ipr = timbi2s_bus_read(bus, TIMBI2S_BUS_IPR); + int i; + + dev_dbg(snd_card_get_device_link(bus->card), + "%s: Entry, bus: %d, IPR %x\n", __func__, BUS_INDEX(bus), ipr); + + for (i = 0; i < NUM_SAMPLES; i++) { + *(u32 *)(runtime->dma_area + bus->buf_pos) = + timbi2s_bus_read(bus, TIMBI2S_BUS_FIFO); + bus->buf_pos += SAMPLE_SIZE; + bus->buf_pos %= buffer_size; + } + + timbi2s_bus_write(bus, ipr, TIMBI2S_BUS_ICLR); + + /* inform ALSA that a period was received */ + snd_pcm_period_elapsed(bus->substream); +} + +static void timbi2s_irq_process_tx(struct timbi2s_bus *bus) +{ + struct snd_pcm_runtime *runtime = bus->substream->runtime; + u32 buffer_size = snd_pcm_lib_buffer_bytes(bus->substream); + u32 ipr = timbi2s_bus_read(bus, TIMBI2S_BUS_IPR); + int i; + + dev_dbg(snd_card_get_device_link(bus->card), + "%s: Entry, bus: %d, IPR %x\n", __func__, BUS_INDEX(bus), ipr); + + for (i = 0; i < NUM_SAMPLES; i++) { + timbi2s_bus_write(bus, + *(u32 *)(runtime->dma_area + bus->buf_pos), + TIMBI2S_BUS_FIFO); + bus->buf_pos += SAMPLE_SIZE; + bus->buf_pos %= buffer_size; + } + + dev_dbg(snd_card_get_device_link(bus->card), "%s: ISR: %x, ICOR: %x\n", + __func__, timbi2s_bus_read(bus, TIMBI2S_BUS_ISR), + timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR)); + + timbi2s_bus_write(bus, ipr, TIMBI2S_BUS_ICLR); + + /* inform ALSA that a period was received */ + snd_pcm_period_elapsed(bus->substream); +} + +static void timbi2s_tasklet(unsigned long arg) +{ + struct snd_card *card = (struct snd_card *)arg; + struct timbi2s *i2s = snd_pcm_chip(card); + u32 uir = ioread32(i2s->membase + TIMBI2S_REG_UIR); + unsigned i; + + dev_dbg(snd_card_get_device_link(card), "%s: Entry, UIR %x\n", + __func__, uir); + + for (i = 0; i < i2s->num_busses; i++) + if (uir & (1 << i)) { + struct timbi2s_bus *bus = i2s->busses + i; + if (BUS_IS_RX(bus)) + timbi2s_irq_process_rx(bus); + else + timbi2s_irq_process_tx(bus); + } + + enable_irq(i2s->irq); +} + +static irqreturn_t timbi2s_irq(int irq, void *devid) +{ + struct timbi2s *i2s = devid; + + tasklet_schedule(&i2s->tasklet); + disable_irq_nosync(i2s->irq); + + return IRQ_HANDLED; +} + +static int timbi2s_setup_busses(struct snd_card *card, + struct platform_device *pdev) +{ + const struct timbi2s_platform_data *pdata = pdev->dev.platform_data; + unsigned i; + + dev_dbg(&pdev->dev, "%s: Entry, no busses: %d, busses: %p\n", __func__, + pdata->num_busses, pdata->busses); + + for (i = 0; i < pdata->num_busses; i++) { + int capture = pdata->busses[i].rx; + int err; + u32 ctl; + struct timbi2s *i2s = snd_pcm_chip(card); + struct timbi2s_bus *bus = i2s->busses + i; + + dev_dbg(&pdev->dev, "%s: Setting up bus: %d\n", __func__, i); + + SET_BUS_INDEX(bus, i); + bus->sample_rate = pdata->busses[i].sample_rate; + bus->card = card; + /* prescaling only applies to master busses, we use the + * knowledge of that to identify the direction later + * eg, bus->prescale != 0 -> master bus + */ + if (capture) + SET_BUS_RX(bus); + + spin_lock_init(&bus->lock); + + if (bus->sample_rate != 44100 && bus->sample_rate != 8000) { + dev_err(&pdev->dev, + "Unsupported bitrate: %d\n", bus->sample_rate); + return -EINVAL; + } + + dev_dbg(&pdev->dev, "%s: Will check HW direction on bus: %d\n", + __func__, BUS_INDEX(bus)); + + /* check that the HW agrees with the direction */ + ctl = timbi2s_bus_read(bus, TIMBI2S_BUS_ICOR); + if ((capture && !(ctl & TIMBI2S_ICOR_RX_ID)) || + (!capture && !(ctl & TIMBI2S_ICOR_TX_ID))) { + dev_dbg(&pdev->dev, + "HW and platform data disagree on direction\n"); + return -EINVAL; + } + + dev_dbg(&pdev->dev, "%s: Will create PCM channel for bus: %d\n", + __func__, BUS_INDEX(bus)); + err = snd_pcm_new(card, card->shortname, i, !capture, + capture, &bus->pcm); + if (err) { + dev_dbg(&pdev->dev, "%s, Failed to create pcm: %d\n", + __func__, err); + return err; + } + + if (capture) + snd_pcm_set_ops(bus->pcm, SNDRV_PCM_STREAM_CAPTURE, + &timbi2s_capture_ops); + if (!capture) + snd_pcm_set_ops(bus->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &timbi2s_playback_ops); + + dev_dbg(&pdev->dev, "%s: Will preallocate buffers to bus: %d\n", + __func__, BUS_INDEX(bus)); + + err = snd_pcm_lib_preallocate_pages_for_all(bus->pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + NUM_SAMPLES * NUM_PERIODS * SAMPLE_SIZE * 2, + NUM_SAMPLES * NUM_PERIODS * SAMPLE_SIZE * 2); + if (err) { + dev_dbg(&pdev->dev, "%s, Failed to create pcm: %d\n", + __func__, err); + + return err; + } + + bus->pcm->private_data = bus; + bus->pcm->info_flags = 0; + strcpy(bus->pcm->name, card->shortname); + i2s->num_busses++; + } + + return 0; +} + +static int __devinit timbi2s_probe(struct platform_device *pdev) +{ + int err; + int irq; + struct timbi2s *i2s; + struct resource *iomem; + const struct timbi2s_platform_data *pdata = pdev->dev.platform_data; + struct snd_card *card; + u32 ver; + + if (!pdata) { + err = -ENODEV; + goto out; + } + + if (pdata->num_busses > MAX_BUSSES) { + err = -EINVAL; + goto out; + } + + iomem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iomem) { + err = -ENODEV; + goto out; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + err = -ENODEV; + goto out; + } + + err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct timbi2s) + + sizeof(struct timbi2s_bus) * pdata->num_busses, &card); + if (err) + goto out; + + strcpy(card->driver, "Timberdale I2S"); + strcpy(card->shortname, "Timberdale I2S"); + sprintf(card->longname, "Timberdale I2S Driver"); + + snd_card_set_dev(card, &pdev->dev); + + i2s = snd_pcm_chip(card); + + if (!request_mem_region(iomem->start, resource_size(iomem), + DRIVER_NAME)) { + err = -EBUSY; + goto err_region; + } + + i2s->membase = ioremap(iomem->start, resource_size(iomem)); + if (!i2s->membase) { + err = -ENOMEM; + goto err_ioremap; + } + + err = timbi2s_setup_busses(card, pdev); + if (err) + goto err_setup; + + tasklet_init(&i2s->tasklet, timbi2s_tasklet, (unsigned long)card); + i2s->irq = irq; + i2s->main_clk = pdata->main_clk; + + err = request_irq(irq, timbi2s_irq, 0, DRIVER_NAME, i2s); + if (err) + goto err_request_irq; + + err = snd_card_register(card); + if (err) + goto err_register; + + platform_set_drvdata(pdev, card); + + ver = ioread32(i2s->membase + TIMBI2S_REG_VER); + + printk(KERN_INFO + "Driver for Timberdale I2S (ver: %d.%d) successfully probed.\n", + ver >> 16 , ver & 0xffff); + + return 0; + +err_register: + free_irq(irq, card); +err_request_irq: +err_setup: + iounmap(i2s->membase); +err_ioremap: + release_mem_region(iomem->start, resource_size(iomem)); +err_region: + snd_card_free(card); +out: + printk(KERN_ERR DRIVER_NAME": Failed to register: %d\n", err); + + return err; +} + +static int __devexit timbi2s_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct timbi2s *i2s = snd_pcm_chip(card); + struct resource *iomem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + tasklet_kill(&i2s->tasklet); + free_irq(i2s->irq, i2s); + + iounmap(i2s->membase); + release_mem_region(iomem->start, resource_size(iomem)); + snd_card_free(card); + + platform_set_drvdata(pdev, 0); + return 0; +} + +static struct platform_driver timbi2s_platform_driver = { + .driver = { + .name = DRIVER_NAME, + .owner = THIS_MODULE, + }, + .probe = timbi2s_probe, + .remove = __devexit_p(timbi2s_remove), +}; + +/*--------------------------------------------------------------------------*/ + +static int __init timbi2s_init(void) +{ + return platform_driver_register(&timbi2s_platform_driver); +} + +static void __exit timbi2s_exit(void) +{ + platform_driver_unregister(&timbi2s_platform_driver); +} + +module_init(timbi2s_init); +module_exit(timbi2s_exit); + +MODULE_ALIAS("platform:"DRIVER_NAME); +MODULE_DESCRIPTION("Timberdale I2S bus driver"); +MODULE_AUTHOR("Mocean Laboratories <info(a)mocean-labs.com>"); +MODULE_LICENSE("GPL v2");

3 6

[alsa-devel] Support for floating point controls?
by Tobias Schneider 01 Feb '10

01 Feb '10

I was searching for floating point controls and found your post. I would also appreciate such a define, but I also found an answer that might still be "up to date" http://thread.gmane.org/gmane.linux.alsa.devel/1498/focus=1507

1 0

[alsa-devel] hw_ptr_interrupt removal broke interrupt pointer updates
by Clemens Ladisch 01 Feb '10

01 Feb '10

Commit "cleanup & merge hw_ptr update functions" says: > The main change is hw_ptr_interrupt variable removal to simplify code > logic. This variable can be computed directly from hw_ptr. The hw_ptr_interrupt variable was needed to differentiate between the position at the last normal pointer update and the position of the last signaled period boundary. if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data. Best regards, Clemens

4 8

[alsa-devel] [PATCH] ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used
by Guennadi Liakhovetski 01 Feb '10

01 Feb '10

In case, if OPCLK is not used, and PLL is used for driving the codec, the choice of PLL output frequency could result in a needlessly imprecise system clock frequency. Use an iterative process to select a precise configuration. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 6d807b1..61b0925 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -58,6 +58,7 @@ struct wm8978_priv { unsigned int f_mclk; unsigned int f_256fs; unsigned int f_opclk; + int mclk_idx; enum wm8978_sysclk_src sysclk; u16 reg_cache[WM8978_CACHEREGNUM]; }; @@ -402,6 +403,35 @@ static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, pll_div->k = k; } + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * find index >= idx, such that, for a given f_out, + * 3 * f_mclk / 4 <= f_PLLOUT < 13 * f_mclk / 4 + * f_out can be f_256fs or f_opclk, currently only used for f_256fs. Can be + * generalised for f_opclk with suitable coefficient arrays, but currently + * the OPCLK divisor is calculated directly, not iteratively. + */ +static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk, + unsigned int *f_pllout) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + unsigned int f_pllout_x4 = 4 * f_out * mclk_numerator[i] / + mclk_denominator[i]; + if (3 * f_mclk <= f_pllout_x4 && f_pllout_x4 < 13 * f_mclk) { + *f_pllout = f_pllout_x4 / 4; + return i; + } + } + + return -EINVAL; +} + /* * Calculate internal frequencies and dividers, according to Figure 40 * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 @@ -412,12 +442,16 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) struct wm8978_pll_div pll_div; unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk, f_256fs = wm8978->f_256fs; - unsigned int f2, opclk_div; + unsigned int f2; if (!f_mclk) return -EINVAL; if (f_opclk) { + unsigned int opclk_div; + /* Cannot set up MCLK divider now, do later */ + wm8978->mclk_idx = -1; + /* * The user needs OPCLK. Choose OPCLKDIV to put * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. @@ -444,7 +478,7 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) wm8978->f_pllout = f_opclk * opclk_div; } else if (f_256fs) { /* - * Not using OPCLK, choose R: + * Not using OPCLK, but PLL is used for the codec, choose R: * 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12. * f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where * prescale = 1, or prescale = 2. Prescale is calculated inside @@ -453,18 +487,11 @@ static int wm8978_configure_pll(struct snd_soc_codec *codec) * f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK * must be 3.781MHz <= f_MCLK <= 32.768MHz */ - if (48 * f_256fs < 3 * f_mclk || 4 * f_256fs >= 13 * f_mclk) - return -EINVAL; + int idx = wm8978_enum_mclk(f_256fs, f_mclk, &wm8978->f_pllout); + if (idx < 0) + return idx; - /* - * MCLKDIV will be selected in .hw_params(), just choose a - * suitable f_PLLOUT - */ - if (4 * f_256fs < 3 * f_mclk) - /* Will have to use MCLKDIV */ - wm8978->f_pllout = wm8978->f_mclk * 3 / 4; - else - wm8978->f_pllout = f_256fs; + wm8978->mclk_idx = idx; /* GPIO1 into default mode as input - before configuring PLL */ snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); @@ -515,6 +542,20 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8978->f_opclk = div; if (wm8978->f_mclk) + /* + * We know the MCLK frequency, the user has requested + * OPCLK, configure the PLL based on that and start it + * and OPCLK immediately. We will configure PLL to match + * user-requested OPCLK frquency as good as possible. + * In fact, it is likely, that matching the sampling + * rate, when it becomes known, is more important, and + * we will not be reconfiguring PLL then, because we + * must not interrupt OPCLK. But it should be fine, + * because typically the user will request OPCLK to run + * at 256fs or 512fs, and for these cases we will also + * find an exact MCLK divider configuration - it will + * be equal to or double the OPCLK divisor. + */ ret = wm8978_configure_pll(codec); break; case WM8978_BCLKDIV: @@ -640,10 +681,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -/* MCLK dividers */ -static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; -static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; - /* * Set PCM DAI bit size and sample rate. */ @@ -709,9 +746,11 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->f_256fs = params_rate(params) * 256; if (wm8978->sysclk == WM8978_MCLK) { + wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { if (!wm8978->f_pllout) { + /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) return ret; @@ -719,32 +758,34 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, f_sel = wm8978->f_pllout; } - /* - * In some cases it is possible to reconfigure PLL to a higher frequency - * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or - * 512 * fs, so, we should be fine. - */ - if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) - return -EINVAL; + if (wm8978->mclk_idx < 0) { + /* Either MCLK is used directly, or OPCLK is used */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; - for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { - diff = abs(wm8978->f_256fs * 3 - - f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); - if (diff < diff_best) { - diff_best = diff; - best = i; - } + if (diff < diff_best) { + diff_best = diff; + best = i; + } - if (!diff) - break; + if (!diff) + break; + } + } else { + /* OPCLK not used, codec driven by PLL */ + best = wm8978->mclk_idx; + diff = 0; } if (diff) - dev_warn(codec->dev, "Imprecise clock: %u%s\n", - f_sel * mclk_denominator[best] / mclk_numerator[best], - wm8978->sysclk == WM8978_MCLK ? - ", consider using PLL" : ""); + dev_warn(codec->dev, "Imprecise sampling rate: %uHz%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best] / 256, + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, params_format(params), params_rate(params), best);

2 1

[alsa-devel] RME HDSPe Raydat
by Dougall Irving 01 Feb '10

01 Feb '10

I've been trying to get an RME HDSPe Raydat working. I've installed the latest firmware on the card using the Windows firmware uploader. The stock hdspm module in ALSA version 1.0.22.1 doesn't work correctly. The first problem I found is that external ADAT equipment were not able to sync with RME adat outputs. Following some old alsa code posted on another site, I changed the HDSPM_freqReg to 100 MHz (instead of 105) and found that I could get external equipment to sync to the raydat at 48kHz. The second issue I had was the audio output was turning on and off every second. I found if I changed the .params to only use float 32 instead of int 32 and I forced the card to use precise pointer I could get continuous audio but only with larger buffer sizes. I get a quite a few buffer underruns also. With smaller buffer sizes I get silence. There is a comment in the code which don't follow. I wonder if there is an issue with this code.... /* hwpointer comes in bytes and is 64Bytes accurate (by docu since PCI Burst) i have experimented that it is at most 64 Byte to much for playing so substraction of 64 byte should be ok for ALSA, but use it only for application where you know what you do since if you come to near with record pointer it can be a disaster */ position &= HDSPM_BufferPositionMask; position = ((position - 64) % (2 * hdspm->period_bytes)) / 4; Does anyone have any suggestions or ideas for debugging these problems? Is there any information RME from available? Alternatively, has anyone had success with this card and earlier firmware revisions?

1 0

[alsa-devel] Some installation questions
by Klaus Schulz 01 Feb '10

01 Feb '10

Hi there. I am still maintaining the (inoffical) Ubuntu Alsa Upgrade Script over at http://ubuntuforums.org/showthread.php?t=1046137 Perhaps somebody can give me a hint on some installation problems I am running into: 1. Quite some people reporting that /proc/asound/version is still showing the "original" revision after running the upgrade-script instead of 1.0.22.1. What could be the issue? What is setting /proc/asound/version? 2. I also introduced a kind of restore function. What it does is basically reinstalling all the required Ubuntu packages from the repo to overwrite the (by my script) upgraded files again. a. My first problem was that I was only able to get the original modules activated when running the module assistant. ( module-assistant -f -t -q a-i alsa-source ) Just a re-installation of kernel packages and alsa-packages didn't work. b. When redoing the Alsa Upgrade after the restore (by my script) , I still see in /proc/asound the old Alsa version. Looks again a bit like I am having slight problems with the module handling FYI: Below I run after the "make install" of the driver package The way the .ko files are handled is that they are copied to /lib/modules/`uname -r`/kernel/sound/ ##################### cd ${ALSASRCDIR}/${DRIVER}/ find ./ -name ''*.ko'' > /tmp/alsa_modules header "Copy modules to target directories..." if [ "`uname -a| grep "2.6.24"`" != "" ] ; then tar -cv -T /tmp/alsa_modules -f /lib/modules/`uname -r`/ubuntu/sound/alsa-driver/${DRIVER}.tar cd /lib/modules/`uname -r`/ubuntu/sound/alsa-driver else tar -cv -T /tmp/alsa_modules -f /lib/modules/`uname -r`/kernel/sound/${DRIVER}.tar cd /lib/modules/`uname -r`/kernel/sound/ fi #Extract new modules, overwriting old ones tar -xvf ${DRIVER}.tar rm *.tar depmod -a ############################## I obviously lack some kind of understanding how all this works. Perhaps somebody can show me directions how to tackle the subject. THX \Klaus

2 1

[alsa-devel] No sound with DH55TC motherboard
by Fabienne Ducroquet 31 Jan '10

31 Jan '10

Hi, I recently bought an Intel DH55TC motherboard, and I can't get the sound card to work. The sound card is: 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) with a Realtek ALC888S codec chip. Output from dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 28 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: ALC888: BIOS auto-probing. autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=1 (0x15/0x0/0x0/0x0/0x0) hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 realtek: No valid SSID, checking pincfg 0x4016c629 for NID 0x1d realtek: Enabling init ASM_ID=0xc629 CODEC_ID=10ec0888 The snd_hda_codec_intelhdmi and snd_hda_codec_realtek modules are loaded, but there is no /proc/asound/modules file and any program trying to use sound says there is no sound card. I tried all of the models listed under ALC888 in HD-Audio-Models.txt, but none of them worked. I tried with a 2.6.32 kernel from Debian, as well as with a 2.6.33-rc6 kernel and with the sound-2.6 tree from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git pulled on top of a 2.6.33-rc6 kernel, and the results were the same in all three cases. If I compile snd_hda_intel in the kernel, I get the lines: ALSA device list: #0: HDA Intel at 0xfe420000 irq 28 after the lines from HDA quoted above in dmesg, while when it's compiled as a module I get: ALSA device list: No soundcards found before those lines, but apart from that the behavior is the same. Attached is the output from alsa-info.sh. Regards, Fabienne

1 0

[alsa-devel] Warning/error from ALSA’s pcm_min.c example. Possible problem?
by Louise Hoffman 31 Jan '10

31 Jan '10

Dear ALSA developers, When I compile ALSA's pcm_min.c[0] example with gcc -Wall -lasound pcm_min.c -o pcm_min Everything is fine, but running it, I get the white noise as expected, but I also get this warning/error: Short write (expected 16384, wrote 7616) Which comes from the last if-statement. One bug is in this line frames = snd_pcm_writei(handle, buffer, sizeof(buffer)); where "sizeof(buffer)" should be "number of frames in the buffer". But there are still some strange things going on. I still get those errors, and if I remove the for-loop, nothing is played. Here is my code. [1] Can someone see what's wrong? Hugs, Louise [0] http://www.google.com/codesearch/p?hl=en&sa=N&cd=1&ct=rc#4FSOSMZ6Pxc/distfi… [1] http://pastebin.com/m2f28b578

3 4

[alsa-devel] [Q] playing / recording mono on a stereo-only hardware
by Guennadi Liakhovetski 30 Jan '10

30 Jan '10

Hi I know this question belongs rather to alsa-users, and I did ask it there, but got no solution so far. Maybe someone here can help. The recently posted SuperH SIU driver only supports stereo playback and capture (channels_min = channels_max = 2). Is there a way to tell alsa-lib to duplicate the mono-channel during playback and to mix recorded channels (or use one of them), if mono recording is requested? Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

2 2

[alsa-devel] [PATCH 5/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

5/5 Patchin' patcher: This patch updates alsa-driver echoaudio.patch . Short description: This patch updates alsa-driver echoaudio.patch . Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch__orig 2009-12-28 16:15:54.000000000 +0100 +++ alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch 2010-01-29 23:35:16.000000000 +0100 @@ -1,6 +1,6 @@ ---- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2006-08-05 14:41:12.000000000 +0200 -+++ echoaudio.c 2006-10-07 20:30:53.000000000 +0200 -@@ -1926,6 +1926,7 @@ +--- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:17:31.000000000 +0100 ++++ echoaudio.c 2010-01-29 23:28:45.000000000 +0100 +@@ -1943,6 +1943,7 @@ static __devinit int snd_echo_create(str struct echoaudio *chip; int err; size_t sz; @@ -8,7 +8,7 @@ static struct snd_device_ops ops = { .dev_free = snd_echo_dev_free, }; -@@ -1988,7 +1989,12 @@ +@@ -2014,7 +2015,12 @@ static __devinit int snd_echo_create(str chip->comm_page_phys = chip->commpage_dma_buf.addr; chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; @@ -19,6 +19,28 @@ + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); +#endif + err = init_hw(chip, chip->pci->device, subsystem_device); - if (err) { - DE_INIT(("init_hw err=%d\n", err)); - snd_echo_free(chip); + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { +@@ -2245,6 +2251,7 @@ static int snd_echo_resume(struct pci_de + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; ++ u16 subsystem_device; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); +@@ -2252,7 +2259,12 @@ static int snd_echo_resume(struct pci_de + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + +- err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); ++#if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 3, 0) ++ subsystem_device = pci->subsystem_device; ++#else ++ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); ++#endif ++ err = init_hw(chip, chip->pci->device, subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); -- Giuliano.

1 0

[alsa-devel] [PATCH 4/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

4/5 Add suspend support #2: This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). P.S. I'll fix "rettings" later. Short description: This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:11:43.000000000 +0100 @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_su spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_su err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(str return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(str chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(str snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(stru snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; Only in alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/: echoaudio.c.orig diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudi if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echo static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct ech chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct ech chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct ech -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:11:43.000000000 +0100 @@ -472,6 +472,8 @@ static void free_firmware(const struct f #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); -- Giuliano.

1 0

[alsa-devel] [PATCH 3/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

3/5 Add suspend support #1: This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Short description: Move the controls init code outside the init_hw() function because is must not be called during resume. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chi chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chi +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chi ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoau control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chi ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chi else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- Giuliano.

1 0

[alsa-devel] [PATCH 2/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

2/4 Add firmware cache #2: This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Short description: This patch implements a simple cache for the firmware files when CONFIG_PM is defined. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 @@ -43,12 +43,24 @@ static int get_firmware(const struct fir { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct fir static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudi pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- Giuliano.

1 0

[alsa-devel] [PATCH 1/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 30 Jan '10

30 Jan '10

1/5 Add firmware cache #1: When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Short description: Changes the way the firmware is passed through functions. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/darla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_3g.c 2010-01-02 21:27:05.000000000 +0100 @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *c /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-02 21:27:05.000000000 +0100 @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOC static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct fir return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -175,15 +175,15 @@ static inline int check_asic_status(stru #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struc return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudi chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/echoaudio.h 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-02 21:27:05.000000000 +0100 @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudi static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/gina24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chi /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *c /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigo_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla20_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *c return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/layla24_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -32,8 +32,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *c /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoau /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct e { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mia_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chi chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff -dup alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig/alsa-kernel/pci/echoaudio/mona_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-02 21:27:05.000000000 +0100 @@ -33,8 +33,7 @@ static int write_control_reg(struct echo static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chi /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *c { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *c mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *c /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC n if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoau if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct e case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- Giuliano.

1 0

[alsa-devel] A driver for Zoom R16 (USB)?
by Jani Frilander 30 Jan '10

30 Jan '10

Hi. I have a Zoom R16 multitrack recorder / USB audio interface / control surface unit. Unfortunately it's audio interface/control surface part currently don't work with linux. When I plug it in kernel (2.6.31-9-rt) won't recognize it as a audio interface. Is there a way to make it work? It probably needs to be hacked, because Zoom won't provide any technical information, according to their email I received. I'm a total newbie in driver development, but I'll be willing to work with this. Regards, Jani Frilander

1 0

[alsa-devel] overrun in second capture device after stopping first capture device
by Tobias Schneider 29 Jan '10

29 Jan '10

Dear ALSA Dev-Team, while testing my driver, I figured out that under some circumstances there occur overruns when recording two streams: first I startet a long capture at device 1: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 121.536 ms long) overrun!!! (at least 115.361 ms long) overrun!!! (at least 190.883 ms long) overrun!!! (at least 121.772 ms long) overrun!!! (at least 119.487 ms long) In another terminal, I started a second capture on the second device. Everytime this capture is done, I got an overrun in the first capture (see above) -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# It seems, that the duration of the overrun relates to the capture time of the second capture, as it increases with bigger capture times. But if capture times are bigger than 25s, it seems there is no overrun...just some logs to clarify: First Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 99.074 ms long) Second Terminal: -bash-3.2# -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 30 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 15 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# First Terminal: -bash-3.2# arecord -D hw:0,1 -d 180 -r 32000 -c 1 -t wav -f S16_LE arecord_049.wav Recording WAVE 'arecord_049.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono overrun!!! (at least 193.656 ms long) Second Terminal: -bash-3.2# arecord -D hw:0,0 -d 25 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# arecord -D hw:0,0 -d 20 -r 32000 -c 1 -t wav -f S16_LE arecord_047.wav Recording WAVE 'arecord_047.wav' : Signed 16 bit Little Endian, Rate 32000 Hz, Mono -bash-3.2# So I am wondering why such an overrun occurs, can someone please give me some hints what my be the reason? Thanks in Advance! Tobias

1 2

[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 29 Jan '10

29 Jan '10

The following changes since commit 64a028a6de08545a2c94f302bc7694bf48aee5b5: Linus Torvalds (1): Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs-2.6 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Pull request against Linus' current master branch since fix/asoc doesn't seem to have been updated - should I be looking at a different branch? Anuj Aggarwal (2): ASoC: AIC23: Fixing writes to non-existing registers in resume function ASoC: AM3517: ASoC driver not getting compiled sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/Makefile | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 29 Jan '10

29 Jan '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 63 +++++ arch/arm/plat-omap/mcbsp.c | 402 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 187 ++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 655 insertions(+), 1 deletions(-)

5 7

[alsa-devel] [PATCH v2] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v1 -> v2: 1. Added " Switch" to wm8974 and wm8978 DAC/ADC oversampling controls sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..ee637af 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..ec2624b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

2 1

[alsa-devel] AT91 AC97 Support broken in menuconfig.
by javier Martin 29 Jan '10

29 Jan '10

Hello, I was trying to get AC97 support for an AT91SAM9G45 in kernel 2.6.32 and I have noticed that it requires that AVR32 is selected, however the following commit states that driver for AVR32 was adapted for AT91 too: commit d9fbd9a2cd5ac1b286a7d7cdb1a180ce1edaee2f Merge: f25f60b... baea7b9... Author: Linus Torvalds <torvalds(a)linux-foundation.org> Date: Thu Sep 24 17:06:51 2009 -0700 Merge branch 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm * 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (103 commits) .... ARM: 5710/1: at91: add AC97 support to at91sam9rl and at91sam9rlek board ARM: 5709/1: at91: add AC97 support to at91sam9g45 series and at91sam9m10g45ek board ARM: 5621/1: at91/dmaengine: integration of at_hdmac driver in at91sam9g45 series ARM: 5620/1: at91/dmaengine: integration of at_hdmac driver in at91sam9rl ... I just wanted to know if this is just an error in Kconfig tree or the driver under "sound/atmel/" does not support at91sam9g54. Thanks. -- Javier Martin Vista Silicon S.L. CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

1 0

[alsa-devel] [PATCH] ASoC: remove bogus SLEEP mode from wm8978 driver
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978 affects codec clocks. Being useless for suspend / resume, it cannot be used in bias-level control either. Remove this bit handling. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8978.c | 7 +------ 1 files changed, 1 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index b86ef47..8dcebaa 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -873,9 +873,7 @@ static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); - /* Put to sleep */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x40); - + return 0; } @@ -887,9 +885,6 @@ static int wm8978_resume(struct platform_device *pdev) int i; u16 *cache = codec->reg_cache; - /* Wake up the codec */ - snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); - /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { if (i == WM8978_RESET) -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: clean up wm8974 and wm8978 clock divider handling
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their .set_clkdiv() methods, which is wrong, because these are simple boolean switches and not clock dividers. Move these bits to sound controls. Also remove manual configuration of the MCLK divider in wm8978, since it is configured automatically. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- sound/soc/codecs/wm8974.c | 12 ++++-------- sound/soc/codecs/wm8974.h | 12 +----------- sound/soc/codecs/wm8978.c | 19 ++++--------------- sound/soc/codecs/wm8978.h | 3 --- sound/soc/sh/migor.c | 4 ---- 5 files changed, 9 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 8812751..051d58d 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode), SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), + +/* DAC / ADC oversampling */ +SOC_SINGLE("DAC 128x Oversampling", WM8974_DAC, 8, 1, 0), +SOC_SINGLE("ADC 128x Oversampling", WM8974_ADC, 8, 1, 0), }; /* Speaker Output Mixer */ @@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; snd_soc_write(codec, WM8974_CLOCK, reg | div); break; - case WM8974_ADCCLK: - reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; - snd_soc_write(codec, WM8974_ADC, reg | div); - break; - case WM8974_DACCLK: - reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; - snd_soc_write(codec, WM8974_DAC, reg | div); - break; case WM8974_BCLKDIV: reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; snd_soc_write(codec, WM8974_CLOCK, reg | div); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h index 98de956..896a7f0 100644 --- a/sound/soc/codecs/wm8974.h +++ b/sound/soc/codecs/wm8974.h @@ -57,17 +57,7 @@ /* Clock divider Id's */ #define WM8974_OPCLKDIV 0 #define WM8974_MCLKDIV 1 -#define WM8974_ADCCLK 2 -#define WM8974_DACCLK 3 -#define WM8974_BCLKDIV 4 - -/* DAC clock dividers */ -#define WM8974_DACCLK_F2 (1 << 3) -#define WM8974_DACCLK_F4 (0 << 3) - -/* ADC clock dividers */ -#define WM8974_ADCCLK_F2 (1 << 3) -#define WM8974_ADCCLK_F4 (0 << 3) +#define WM8974_BCLKDIV 2 /* PLL Out dividers */ #define WM8974_OPCLKDIV_1 (0 << 4) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 8dcebaa..6d807b1 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -210,6 +210,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { /* Speaker */ SOC_DOUBLE_R("Speaker Switch", WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), + + /* DAC / ADC oversampling */ + SOC_SINGLE("DAC 128x Oversampling", WM8978_DAC_CONTROL, 8, 1, 0), + SOC_SINGLE("ADC 128x Oversampling", WM8978_ADC_CONTROL, 8, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ @@ -513,21 +517,6 @@ static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, if (wm8978->f_mclk) ret = wm8978_configure_pll(codec); break; - case WM8978_MCLKDIV: - if (div & ~0xe0) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); - break; - case WM8978_ADCCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); - break; - case WM8978_DACCLK: - if (div & ~8) - return -EINVAL; - snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); - break; case WM8978_BCLKDIV: if (div & ~0x1c) return -EINVAL; diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index b58f0bf..56ec832 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -72,9 +72,6 @@ /* Clock divider Id's */ enum wm8978_clk_id { WM8978_OPCLKRATE, - WM8978_MCLKDIV, - WM8978_ADCCLK, - WM8978_DACCLK, WM8978_BCLKDIV, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 3ccd9b3..b823a5c 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -59,10 +59,6 @@ static int migor_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); if (ret < 0) return ret; -- 1.6.2.4

2 1

[alsa-devel] [PATCH 2b/4 v2] ASoC: add support for the sh7722 Migo-R board
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v2 -> v3: 1. adjusted to the new wm8978 clocking scheme 2. added .hw_free() to switch off PLL diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d..a86696b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -55,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0a..8a5a192 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 0000000..3ccd9b3 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/module.h> + +#include <asm/clock.h> + +#include <cpu/sh7722.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2");

2 1

[alsa-devel] [PATCH 1/1] ASoC: AM3517: ASoC driver not getting compiled
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..19283e5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

2 1

[alsa-devel] [PATCH] ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 23 +++++++---------------- 1 files changed, 7 insertions(+), 16 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d..24fd39f 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- 1.6.2.5

3 2

[alsa-devel] [PATCH 1/1] ASoC: AIC23: Fixing writes to non-existing registers in resume function
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb..da589d8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: Modify codec default settings
by Peter Ujfalusi 29 Jan '10

29 Jan '10

Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 74f0d65..e0106a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,12 +64,12 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ @@ -99,7 +99,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ - 0x16, /* REG_APLL_CTL (0x3A) */ + 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ -- 1.6.6.1

3 5

[alsa-devel] [PATCHv2 05/10] ASoC: AC97: SMDK: Add wm9713 machine driver
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> This patch adds the common machine driver for SMDKs that have a WM9713 codec attched to the AC97 controller. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 +++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk_wm9713.c | 97 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 107 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk_wm9713.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index ad3690e..d1c6f93 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -115,3 +115,11 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b7411bd..1117678 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c new file mode 100644 index 0000000..7dd933f --- /dev/null +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -0,0 +1,97 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/soc.h> + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + Playback (HeadPhone):- + Headphone Playback Switch - On + $ amixer cset numid=4 1 + + Right Headphone Out Mux - Headphone + $ amixer cset numid=92 2 + Left Headphone Out Mux - Headphone + $ amixer cset numid=93 2 + + Right HP Mixer PCM Playback Switch - On + $ amixer cset numid=75 1 + Left HP Mixer PCM Playback Switch - On + $ amixer cset numid=81 1 + + Capture (LineIn):- + Right Capture Source - Line + $ amixer cset numid=86 2 + Left Capture Source - Line + $ amixer cset numid=87 2 +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

3 3

[alsa-devel] [PATCH - hda intel conextant 1/1] Fixed microphone mixer for HP Compaq Presario F700 Added conextant patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea
by knprox＠mi.rr.com 28 Jan '10

28 Jan '10

From: Ken Prox <kprox(a)users.sourceforge.net> Signed-off-by: Ken Prox <kprox(a)users.sourceforge.net> --- pci/hda/patch_conexant.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 51 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 947785f..6142d75 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -1723,6 +1723,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_f700_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1813,6 +1829,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_f700_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1832,6 +1874,7 @@ enum { CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ + CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_MODELS }; @@ -1840,6 +1883,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", + [CXT5051_F700] = "hp 700" }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1849,6 +1893,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; @@ -1899,8 +1944,14 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; + case CXT5051_F700: + spec->init_verbs[0] = cxt5051_f700_init_verbs; + spec->mixers[0] = cxt5051_f700_mixers; + spec->no_auto_mic = 1; + break; } + return 0; } -- 1.6.2.5

5 11

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 28 Jan '10

28 Jan '10

The following changes since commit 0d34e91596ef537c2893a031f0483014bb82adf3: Guennadi Liakhovetski (1): ASoC: add a WM8978 codec driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Kuninori Morimoto (1): ASoC: fsi: Add spin lock operation for accessing shared area Mark Brown (3): ASoC: Remove version display from WM9713 ASoC: Add TLV information and additional volumes to WM9713 ASoC: Set codec->dev for AC97 devices Peter Ujfalusi (1): ASoC: TWL4030: Modify codec default settings sound/soc/codecs/twl4030.c | 8 +++--- sound/soc/codecs/wm9713.c | 64 ++++++++++++++++++++++++++++++------------- sound/soc/sh/fsi.c | 29 ++++++++++++++++++-- sound/soc/soc-core.c | 1 + 4 files changed, 75 insertions(+), 27 deletions(-)

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[alsa-devel] [PATCH] ASoC: fsi: Add spin lock operation for accessing shared area
by Kuninori Morimoto 28 Jan '10

28 Jan '10

fsi_master_xxx function should be protected by spin lock, because it are used from both FSI-A and FSI-B. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 29 ++++++++++++++++++++++++++--- 1 files changed, 26 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5f9f269..ebf3588 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -110,6 +110,7 @@ struct fsi_master { struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; + spinlock_t lock; }; /************************************************************************ @@ -168,30 +169,51 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_write((u32)(master->base + reg), data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_write((u32)(master->base + reg), data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) { + u32 ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return 0; - return __fsi_reg_read((u32)(master->base + reg)); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_read((u32)(master->base + reg)); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static int fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } /************************************************************************ @@ -929,6 +951,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsib.base = master->base + 0x40; master->fsib.master = master; + spin_lock_init(&master->lock); pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); -- 1.6.3.3

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[alsa-devel] [PATCH]Change the AZX_MAX_PCMS to 10
by Wei Ni 28 Jan '10

28 Jan '10

Hi, Takashi When I develop hdmi audio driver on the GT21x, I found a bug in the had_intel.c In hda_codec.c, it has define "[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },", it support up to device 9 for HDMI. But in hda_intel.c, it only define AZX_MAX_PCMS as 8. So the GT21x have 4 hdmi codecs, when run azx_attach_pcm_stream(), it will show error "Invalid PCM device number 8", and "... number 9", and return "-EINVAL". We should change the AZX_MAX_PCMS to 10. I attach the patch file, please check it. Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

3 2

[alsa-devel] [PATCHv3 06/10] ARM: SMDK64XX: Enable AC97 device
by jassisinghbrar＠gmail.com 28 Jan '10

28 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Add AC97 controller to platform devices and initialize the AC97 gpios. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/mach-s3c6410/mach-smdk6410.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-s3c6410/mach-smdk6410.c b/arch/arm/mach-s3c6410/mach-smdk6410.c index eba345f..f2c70ec 100644 --- a/arch/arm/mach-s3c6410/mach-smdk6410.c +++ b/arch/arm/mach-s3c6410/mach-smdk6410.c @@ -55,6 +55,7 @@ #include <plat/gpio-cfg.h> #include <plat/s3c6410.h> +#include <plat/audio.h> #include <plat/clock.h> #include <plat/devs.h> #include <plat/cpu.h> @@ -253,7 +254,7 @@ static struct platform_device *smdk6410_devices[] __initdata = { &smdk6410_b_pwr_5v, #endif &smdk6410_lcd_powerdev, - + &s3c64xx_device_ac97, &smdk6410_smsc911x, }; @@ -475,6 +476,8 @@ static void __init smdk6410_machine_init(void) i2c_register_board_info(0, i2c_devs0, ARRAY_SIZE(i2c_devs0)); i2c_register_board_info(1, i2c_devs1, ARRAY_SIZE(i2c_devs1)); + s3c64xx_ac97_setup_gpio(S3C64XX_AC97_GPD); + platform_add_devices(smdk6410_devices, ARRAY_SIZE(smdk6410_devices)); } -- 1.6.2.5

1 0

[alsa-devel] [PATCH 4/7] ASoC: S3C64XX: USE DMA info from platform data
by jassisinghbrar＠gmail.com 28 Jan '10

28 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Instead of hardcoding DMA channel numbers and address, for each controller in the driver, let the information be read from platform data. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c64xx-i2s.c | 31 +++++++++++++++++++++---------- 1 files changed, 21 insertions(+), 10 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 93ed3aa..8ca3c1d 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -147,6 +147,7 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { struct s3c_i2sv2_info *i2s; struct snd_soc_dai *dai; + struct resource *res; int ret; if (pdev->id >= MAX_I2SV3) { @@ -174,17 +175,27 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; - if (pdev->id == 0) { - i2s->dma_capture->channel = DMACH_I2S0_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S0_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; - } else { - i2s->dma_capture->channel = DMACH_I2S1_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S1_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "Unable to get I2S-TX dma resource\n"); + return -ENXIO; } + i2s->dma_playback->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "Unable to get I2S-RX dma resource\n"); + return -ENXIO; + } + i2s->dma_capture->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Unable to get I2S SFR address\n"); + return -ENXIO; + } + i2s->dma_capture->dma_addr = res->start + S3C2412_IISRXD; + i2s->dma_playback->dma_addr = res->start + S3C2412_IISTXD; i2s->dma_capture->client = &s3c64xx_dma_client_in; i2s->dma_capture->dma_size = 4; -- 1.6.2.5

4 3

[alsa-devel] [PATCHv2 06/10] ARM: SMDK64XX: Enable AC97 device
by jassisinghbrar＠gmail.com 28 Jan '10

28 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Add AC97 controller to platform devices and initialize the AC97 gpios. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/mach-s3c6410/mach-smdk6410.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-s3c6410/mach-smdk6410.c b/arch/arm/mach-s3c6410/mach-smdk6410.c index eba345f..259ebbb 100644 --- a/arch/arm/mach-s3c6410/mach-smdk6410.c +++ b/arch/arm/mach-s3c6410/mach-smdk6410.c @@ -55,6 +55,7 @@ #include <plat/gpio-cfg.h> #include <plat/s3c6410.h> +#include <plat/audio.h> #include <plat/clock.h> #include <plat/devs.h> #include <plat/cpu.h> @@ -253,7 +254,9 @@ static struct platform_device *smdk6410_devices[] __initdata = { &smdk6410_b_pwr_5v, #endif &smdk6410_lcd_powerdev, - +#ifdef CONFIG_SND_SOC_SMDK_WM9713 + &s3c64xx_device_ac97, +#endif &smdk6410_smsc911x, }; @@ -475,6 +478,10 @@ static void __init smdk6410_machine_init(void) i2c_register_board_info(0, i2c_devs0, ARRAY_SIZE(i2c_devs0)); i2c_register_board_info(1, i2c_devs1, ARRAY_SIZE(i2c_devs1)); +#ifdef CONFIG_SND_SOC_SMDK_WM9713 + s3c64xx_ac97_setup_gpio(S3C64XX_AC97_GPD); +#endif + platform_add_devices(smdk6410_devices, ARRAY_SIZE(smdk6410_devices)); } -- 1.6.2.5

4 4

[alsa-devel] Synchronizing multiple machines
by Jon Smirl 28 Jan '10

28 Jan '10

What's the best strategy for synchronizing alsa between multiple machines within earshot of each other? I can synchronize the CPU clocks to microsecond accuracy. But how do I deal with drift in the audio clocks? Can I periodically stop the stream and restart it using CPU clock? That would compensate for the audio clock drift. -- Jon Smirl jonsmirl(a)gmail.com

1 0

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 28 Jan '10

28 Jan '10

The following changes since commit f1487fcbe47cd2bc0c71e8598bf9eb6a82dec544: Mark Brown (1): Merge branch 'for-2.6.33' into for-2.6.34 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Barry Song (2): ASoC: ad1836: use soc-cache framework for codec registers access ASoC: ad1938: use soc-cache framework for codec registers access Chaithrika U S (1): ASoC: DaVinci: Fix stream restart error Guennadi Liakhovetski (1): ASoC: add a WM8978 codec driver Jassi Brar (5): ASoC: AC97: S3C: Add controller driver ASoC: AC97: SMDK: Add wm9713 machine driver ASoC: AC97: SMDK2443: Switch to s3c-ac97.c ASoC: AC97: LN2440SBC: Switch to s3c-ac97.c ASoC: AC97: S3C2443: Remove unused driver Mark Brown (1): ASoC: Note jumper settings for smdk_wm9713 driver on SMDK6410 sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 92 +-- sound/soc/codecs/ad1938.c | 164 ++---- sound/soc/codecs/wm8978.c | 1124 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8978.h | 89 +++ sound/soc/davinci/davinci-mcasp.c | 5 +- sound/soc/s3c24xx/Kconfig | 22 +- sound/soc/s3c24xx/Makefile | 7 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 4 +- sound/soc/s3c24xx/s3c-ac97.c | 518 ++++++++++++++++ sound/soc/s3c24xx/s3c-ac97.h | 23 + sound/soc/s3c24xx/s3c2443-ac97.c | 432 ------------- sound/soc/s3c24xx/s3c24xx-ac97.h | 25 - sound/soc/s3c24xx/smdk2443_wm9710.c | 4 +- sound/soc/s3c24xx/smdk_wm9713.c | 103 +++ sound/soc/soc-cache.c | 175 ++++++ 17 files changed, 2123 insertions(+), 670 deletions(-) create mode 100644 sound/soc/codecs/wm8978.c create mode 100644 sound/soc/codecs/wm8978.h create mode 100644 sound/soc/s3c24xx/s3c-ac97.c create mode 100644 sound/soc/s3c24xx/s3c-ac97.h delete mode 100644 sound/soc/s3c24xx/s3c2443-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-ac97.h create mode 100644 sound/soc/s3c24xx/smdk_wm9713.c

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[alsa-devel] [Rev2]Samsung: New driver for AC97 controllers
by jassisinghbrar＠gmail.com 28 Jan '10

28 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Hi, I am going to submit revised patches for new driver for the AC97-ver-2.0 controllers found in Samsung SoCs. The driver is named generic s3c-ac97.c The driver has been tested for on s3c64xx and newer SoCs and is confirmed to support Playback/Capture and VRA. A machine driver, for SMDKs with wm9713 as AC97 codec attached, has also been added. The driver is generic enough to be used as such with newer SoCs. The controller driver patch is meant for Mark Brown's for-2.6.34 Other patches apply cleanly against the Ben Dooks' next-samsung Last few patches move smd2443 and ln2440sbc ac97 drivers to use new s3c-ac97.c and at last the s3c2443-ac97.c is removed. Regards.

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[alsa-devel] I hear sound from headphones but i can't hear sound without headphones
by Angelo D. 27 Jan '10

27 Jan '10

Hi, i have a problem with my audio card. I hear sound from headphones but i can't hear sound without headphones.This is the result of the esecution alsa-info.sh: http://www.alsa-project.org/db/?f=4a72836b153be44d5a3b84bce66638fe41afadb7 This is the link to my related post on ubuntuforum: http://ubuntuforums.org/showthread.php?p=8707917#post8707917 AngeloD

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[alsa-devel] Creative X-Fi Xtreme Audio not working with hda-intel one more
by PChot 27 Jan '10

27 Jan '10

I have the same problems as http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg25644.html my info http://www.alsa-project.org/db/?f=8ff83b31531c6204deddced45d9cf05a1058ac25 what we have to send (which info) that would found some solution Regards, Jan

1 0

[alsa-devel] [PATCH 1/4 v3] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 27 Jan '10

27 Jan '10

The WM8978 codec from Wolfson Microelectronics is very similar to wm8974, but is stereo and also has some differences in pin configuration and internal signal routing. This driver is based on wm8974 and takes the differences into account. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v2 -> v3: 1. converted left-right control pairs to stereo controls 2. moved controls, irrelevant for PM out of DAPM to "usual" controls 3. using snd_soc_update_bits() instead of reading / writing registers 4. removed .set_pll(), codec input frequency and system clock selection are now configured in .set_sysclk(), OPCLK output per .set_clkdiv(OPCLKRATE, f_output), in which case PLL is started early, if OPCLK is not required, but PLL is used, it is configured and started in .hw_params() diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 62ff26a..0aad72f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8974 if I2C + select SND_SOC_WM8978 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM8993 if I2C @@ -230,6 +231,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8974 tristate +config SND_SOC_WM8978 + tristate + config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ea98354..fbd290e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -44,6 +44,7 @@ snd-soc-wm8960-objs := wm8960.o snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8974-objs := wm8974.o +snd-soc-wm8978-objs := wm8978.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8993-objs := wm8993.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o +obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c new file mode 100644 index 0000000..99ce776 --- /dev/null +++ b/sound/soc/codecs/wm8978.c @@ -0,0 +1,1088 @@ +/* + * wm8978.c -- WM8978 ALSA SoC Audio Codec driver + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * Copyright (C) 2007 Carlos Munoz <carlos(a)kenati.com> + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * Based on wm8974 and wm8990 by Liam Girdwood <lrg(a)slimlogic.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <asm/div64.h> + +#include "wm8978.h" + +static struct snd_soc_codec *wm8978_codec; + +/* wm8978 register cache. Note that register 0 is not included in the cache. */ +static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ + 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ + 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ + 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ + 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ + 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ + 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ + 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ + 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ + 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ + 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ + 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ + 0x0001, 0x0001, /* 0x38...0x3b */ +}; + +/* codec private data */ +struct wm8978_priv { + struct snd_soc_codec codec; + unsigned int f_pllout; + unsigned int f_mclk; + unsigned int f_256fs; + unsigned int f_opclk; + enum wm8978_sysclk_src sysclk; + u16 reg_cache[WM8978_CACHEREGNUM]; +}; + +static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"}; +static const char *wm8978_eqmode[] = {"Capture", "Playback"}; +static const char *wm8978_bw[] = {"Narrow", "Wide"}; +static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"}; +static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"}; +static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"}; +static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"}; +static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; +static const char *wm8978_alc3[] = {"ALC", "Limiter"}; +static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; + +static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); + +static const struct snd_kcontrol_new wm8978_snd_controls[] = { + + SOC_SINGLE("Digital Loopback Switch", + WM8978_COMPANDING_CONTROL, 0, 1, 0), + + SOC_ENUM("ADC Companding", adc_compand), + SOC_ENUM("DAC Companding", dac_compand), + + SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("PCM Volume", + WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0), + SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Volume", + WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME, + 0, 255, 0, digital_tlv), + + SOC_ENUM("Equaliser Function", eqmode), + SOC_ENUM("EQ1 Cut Off", eq1), + SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ2 Bandwith", eq2bw), + SOC_ENUM("EQ2 Cut Off", eq2), + SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ3 Bandwith", eq3bw), + SOC_ENUM("EQ3 Cut Off", eq3), + SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv), + + SOC_ENUM("Equaliser EQ4 Bandwith", eq4bw), + SOC_ENUM("EQ4 Cut Off", eq4), + SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv), + + SOC_ENUM("EQ5 Cut Off", eq5), + SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv), + + SOC_SINGLE("DAC Playback Limiter Switch", + WM8978_DAC_LIMITER_1, 8, 1, 0), + SOC_SINGLE("DAC Playback Limiter Decay", + WM8978_DAC_LIMITER_1, 4, 15, 0), + SOC_SINGLE("DAC Playback Limiter Attack", + WM8978_DAC_LIMITER_1, 0, 15, 0), + + SOC_SINGLE("DAC Playback Limiter Threshold", + WM8978_DAC_LIMITER_2, 4, 7, 0), + SOC_SINGLE("DAC Playback Limiter Boost", + WM8978_DAC_LIMITER_2, 0, 15, 0), + + SOC_ENUM("ALC Enable Switch", alc1), + SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), + SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), + + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), + + SOC_ENUM("ALC Capture Mode", alc3), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + + SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), + SOC_SINGLE("ALC Capture Noise Gate Threshold", + WM8978_NOISE_GATE, 0, 7, 0), + + SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 7, 1, 0), + + /* OUT1 - Headphones */ + SOC_DOUBLE_R("Headphone Playback ZC Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT2 - Speakers */ + SOC_DOUBLE_R("Speaker Playback ZC Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0), + + SOC_DOUBLE_R_TLV("Speaker Playback Volume", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, + 0, 63, 0, spk_tlv), + + /* OUT3/4 - Line Output */ + SOC_DOUBLE_R("Line Playback Switch", + WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1), + + /* Mixer #3: Boost (Input) mixer */ + SOC_DOUBLE_R("PGA Boost (+20dB)", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 8, 1, 0), + SOC_DOUBLE_R_TLV("L2/R2 Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 4, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Aux Boost Volume", + WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL, + 0, 7, 0, boost_tlv), + + /* Input PGA volume */ + SOC_DOUBLE_R_TLV("Input PGA Volume", + WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL, + 0, 63, 0, inpga_tlv), + + /* Headphone */ + SOC_DOUBLE_R("Headphone Mute Switch", + WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1), + + /* Speaker */ + SOC_DOUBLE_R("Speaker Mute Switch", + WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), +}; + +/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ +static const struct snd_kcontrol_new wm8978_left_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8978_right_out_mixer[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0), +}; + +/* OUT3/OUT4 Mixer not implemented */ + +/* Mixer #2: Input PGA Mute */ +static const struct snd_kcontrol_new wm8978_left_input_mixer[] = { + SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0), +}; +static const struct snd_kcontrol_new wm8978_right_input_mixer[] = { + SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0), + SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0), + SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + WM8978_POWER_MANAGEMENT_3, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + WM8978_POWER_MANAGEMENT_2, 1, 0), + + /* Mixer #1: OUT1,2 */ + SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3, + 2, 0, wm8978_left_out_mixer), + SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3, + 3, 0, wm8978_right_out_mixer), + + SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2, + 2, 0, wm8978_left_input_mixer), + SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2, + 3, 0, wm8978_right_input_mixer), + + SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL, + 6, 1, NULL, 0), + + SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2, + 8, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3, + 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3, + 8, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0), + + SND_SOC_DAPM_INPUT("LMICN"), + SND_SOC_DAPM_INPUT("LMICP"), + SND_SOC_DAPM_INPUT("RMICN"), + SND_SOC_DAPM_INPUT("RMICP"), + SND_SOC_DAPM_INPUT("LAUX"), + SND_SOC_DAPM_INPUT("RAUX"), + SND_SOC_DAPM_INPUT("L2"), + SND_SOC_DAPM_INPUT("R2"), + SND_SOC_DAPM_OUTPUT("LHP"), + SND_SOC_DAPM_OUTPUT("RHP"), + SND_SOC_DAPM_OUTPUT("LSPK"), + SND_SOC_DAPM_OUTPUT("RSPK"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Output mixer */ + {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, + {"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"}, + + {"Left Output Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left Output Mixer", "Aux Playback Switch", "LAUX"}, + {"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"}, + + /* Outputs */ + {"Right Headphone Out", NULL, "Right Output Mixer"}, + {"RHP", NULL, "Right Headphone Out"}, + + {"Left Headphone Out", NULL, "Left Output Mixer"}, + {"LHP", NULL, "Left Headphone Out"}, + + {"Right Speaker Out", NULL, "Right Output Mixer"}, + {"RSPK", NULL, "Right Speaker Out"}, + + {"Left Speaker Out", NULL, "Left Output Mixer"}, + {"LSPK", NULL, "Left Speaker Out"}, + + /* Boost Mixer */ + {"Right ADC", NULL, "Right Boost Mixer"}, + + {"Right Boost Mixer", NULL, "RAUX"}, + {"Right Boost Mixer", NULL, "Right Capture PGA"}, + {"Right Boost Mixer", NULL, "R2"}, + + {"Left ADC", NULL, "Left Boost Mixer"}, + + {"Left Boost Mixer", NULL, "LAUX"}, + {"Left Boost Mixer", NULL, "Left Capture PGA"}, + {"Left Boost Mixer", NULL, "L2"}, + + /* Input PGA */ + {"Right Capture PGA", NULL, "Right Input Mixer"}, + {"Left Capture PGA", NULL, "Left Input Mixer"}, + + {"Right Input Mixer", "R2 Switch", "R2"}, + {"Right Input Mixer", "MicN Switch", "RMICN"}, + {"Right Input Mixer", "MicP Switch", "RMICP"}, + + {"Left Input Mixer", "L2 Switch", "L2"}, + {"Left Input Mixer", "MicN Switch", "LMICN"}, + {"Left Input Mixer", "MicP Switch", "LMICP"}, +}; + +static int wm8978_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8978_dapm_widgets, + ARRAY_SIZE(wm8978_dapm_widgets)); + + /* set up the WM8978 audio map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* PLL divisors */ +struct wm8978_pll_div { + u32 k; + u8 n; + u8 div2; +}; + +#define FIXED_PLL_SIZE (1 << 24) + +static void pll_factors(struct wm8978_pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 k_part; + unsigned int k, n_div, n_mod; + + n_div = target / source; + if (n_div < 6) { + source >>= 1; + pll_div->div2 = 1; + n_div = target / source; + } else { + pll_div->div2 = 0; + } + + if (n_div < 6 || n_div > 12) + dev_warn(wm8978_codec->dev, + "WM8978 N value exceeds recommended range! N = %u\n", + n_div); + + pll_div->n = n_div; + n_mod = target - source * n_div; + k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2; + + do_div(k_part, source); + + k = k_part & 0xFFFFFFFF; + + pll_div->k = k; +} +/* + * @freq: f_PLLOUT - output PLL frequency + * + * Calculate internal frequencies and dividers, according to Figure 40 + * "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6 + */ +static int wm8978_configure_pll(struct snd_soc_codec *codec) +{ + struct wm8978_priv *wm8978 = codec->private_data; + struct wm8978_pll_div pll_div; + unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk; + unsigned int f2, opclk_div; + + if (!f_mclk) + return -EINVAL; + + if (f_opclk) { + /* + * The user needs OPCLK. Choose OPCLKDIV to put + * 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4. + * f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where + * prescale = 1, or prescale = 2. Prescale is calculated inside + * pll_factors((). We have to select f_PLLOUT, such that + * f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be + * f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4. + */ + if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk) + return -EINVAL; + + if (4 * f_opclk < 3 * f_mclk) + /* Have to use OPCLKDIV */ + opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk; + else + opclk_div = 1; + + dev_dbg(codec->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div); + + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 0x30, + (opclk_div - 1) << 4); + + wm8978->f_pllout = f_opclk * opclk_div; + } else if (wm8978->f_256fs) { + /* Not using OPCLK, just set f_PLLOUT = f_256fs */ + wm8978->f_pllout = wm8978->f_256fs; + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + } else { + return -EINVAL; + } + + f2 = wm8978->f_pllout * 4; + + dev_dbg(codec->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__, + wm8978->f_mclk, wm8978->f_pllout); + + pll_factors(&pll_div, f2, wm8978->f_mclk); + + dev_dbg(codec->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n", + __func__, pll_div.n, pll_div.k, pll_div.div2); + + snd_soc_write(codec, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n); + snd_soc_write(codec, WM8978_PLL_K1, pll_div.k >> 18); + snd_soc_write(codec, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8978_PLL_K3, pll_div.k & 0x1ff); + + /* Turn PLL on */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20); + + if (f_opclk) + /* Output PLL (OPCLK) to GPIO1 */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 4); + + return 0; +} + +/* + * Configure WM8978 clock dividers. + */ +static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + switch (div_id) { + case WM8978_OPCLKRATE: + wm8978->f_opclk = div; + + if (wm8978->f_mclk) + ret = wm8978_configure_pll(codec); + break; + case WM8978_MCLKDIV: + if (div & ~0xe0) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, div); + break; + case WM8978_ADCCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_ADC_CONTROL, 8, div); + break; + case WM8978_DACCLK: + if (div & ~8) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 8, div); + break; + case WM8978_BCLKDIV: + if (div & ~0x1c) + return -EINVAL; + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x1c, div); + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "%s: ID %d, value %u\n", __func__, div_id, div); + + return ret; +} + +/* + * @freq: when .set_pll() us not used, freq is codec MCLK input frequency + */ +static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8978_priv *wm8978 = codec->private_data; + int ret = 0; + + dev_dbg(codec->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq); + + if (freq) { + wm8978->f_mclk = freq; + + /* Even if MCLK is used for system clock, might have to drive OPCLK */ + if (wm8978->f_opclk) + ret = wm8978_configure_pll(codec); + + /* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */ + + if (!ret) + wm8978->sysclk = clk_id; + } + + if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) { + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + + /* GPIO1 into default mode as input - before configuring PLL */ + snd_soc_update_bits(codec, WM8978_GPIO_CONTROL, 7, 0); + + /* Turn off PLL */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, 0x20, 0); + wm8978->sysclk = WM8978_MCLK; + wm8978->f_pllout = 0; + wm8978->f_opclk = 0; + } + + return ret; +} + +/* + * Set ADC and Voice DAC format. + */ +static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + /* + * BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80, + * Data Format mask = 0x18: all will be calculated anew + */ + u16 iface = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x198; + u16 clk = snd_soc_read(codec, WM8978_CLOCKING); + + dev_dbg(codec->dev, "%s\n", __func__); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + clk &= ~1; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x18; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x80; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); + snd_soc_write(codec, WM8978_CLOCKING, clk); + + return 0; +} + +/* MCLK dividers */ +static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12}; +static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1}; + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8978_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8978_priv *wm8978 = codec->private_data; + /* Word length mask = 0x60 */ + u16 iface_ctl = snd_soc_read(codec, WM8978_AUDIO_INTERFACE) & ~0x60; + /* Sampling rate mask = 0xe (for filters) */ + u16 add_ctl = snd_soc_read(codec, WM8978_ADDITIONAL_CONTROL) & ~0xe; + u16 clking = snd_soc_read(codec, WM8978_CLOCKING); + enum wm8978_sysclk_src current_clk_id = clking & 0x100 ? + WM8978_PLL : WM8978_MCLK; + unsigned int f_sel, diff, diff_best = INT_MAX; + int i, best = 0; + + if (!wm8978->f_mclk) + return -EINVAL; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_ctl |= 0x20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_ctl |= 0x40; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_ctl |= 0x60; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case 8000: + add_ctl |= 0x5 << 1; + break; + case 11025: + add_ctl |= 0x4 << 1; + break; + case 16000: + add_ctl |= 0x3 << 1; + break; + case 22050: + add_ctl |= 0x2 << 1; + break; + case 32000: + add_ctl |= 0x1 << 1; + break; + case 44100: + case 48000: + break; + } + + /* Sampling rate is known now, can configure the MCLK divider */ + wm8978->f_256fs = params_rate(params) * 256; + + if (wm8978->sysclk == WM8978_MCLK) { + f_sel = wm8978->f_mclk; + } else { + if (!wm8978->f_pllout) { + int ret = wm8978_configure_pll(codec); + if (ret < 0) + return ret; + } + f_sel = wm8978->f_pllout; + } + + /* + * In some cases it is possible to reconfigure PLL to a higher frequency + * by raising OPCLKDIV, but normally OPCLK is configured to 256 * fs or + * 512 * fs, so, we should be fine. + */ + if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) { + diff = abs(wm8978->f_256fs * 3 - + f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]); + + if (diff < diff_best) { + diff_best = diff; + best = i; + } + + if (!diff) + break; + } + + if (diff) + dev_warn(codec->dev, "Imprecise clock: %u%s\n", + f_sel * mclk_denominator[best] / mclk_numerator[best], + wm8978->sysclk == WM8978_MCLK ? + ", consider using PLL" : ""); + + dev_dbg(codec->dev, "%s: fmt %d, rate %u, MCLK divisor #%d\n", __func__, + params_format(params), params_rate(params), best); + + /* MCLK divisor mask = 0xe0 */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0xe0, best << 5); + + snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface_ctl); + snd_soc_write(codec, WM8978_ADDITIONAL_CONTROL, add_ctl); + + if (wm8978->sysclk != current_clk_id) { + if (wm8978->sysclk == WM8978_PLL) + /* Run CODEC from PLL instead of MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, + 0x100, 0x100); + else + /* Clock CODEC directly from MCLK */ + snd_soc_update_bits(codec, WM8978_CLOCKING, 0x100, 0); + } + + return 0; +} + +static int wm8978_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s: %d\n", __func__, mute); + + if (mute) + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0x40); + else + snd_soc_update_bits(codec, WM8978_DAC_CONTROL, 0x40, 0); + + return 0; +} + +static int wm8978_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 1; /* VMID 75k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_STANDBY: + /* bit 3: enable bias, bit 2: enable I/O tie off buffer */ + power1 |= 0xc; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, + power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); + break; + case SND_SOC_BIAS_OFF: + /* Preserve PLL - OPCLK may be used by someone */ + snd_soc_update_bits(codec, WM8978_POWER_MANAGEMENT_1, ~0x20, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0); + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0); + break; + } + + dev_dbg(codec->dev, "%s: %d, %x\n", __func__, level, power1); + + codec->bias_level = level; + return 0; +} + +#define WM8978_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8978_dai_ops = { + .hw_params = wm8978_hw_params, + .digital_mute = wm8978_mute, + .set_fmt = wm8978_set_dai_fmt, + .set_clkdiv = wm8978_set_dai_clkdiv, + .set_sysclk = wm8978_set_dai_sysclk, +}; + +/* Also supports 12kHz */ +struct snd_soc_dai wm8978_dai = { + .name = "WM8978 HiFi", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = WM8978_FORMATS, + }, + .ops = &wm8978_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8978_dai); + +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8978_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { + if (i == WM8978_RESET) + continue; + if (cache[i] != wm8978_reg[i]) + snd_soc_write(codec, i, cache[i]); + } + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int wm8978_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8978_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8978_codec; + codec = wm8978_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8978_snd_controls, + ARRAY_SIZE(wm8978_snd_controls)); + wm8978_add_widgets(codec); + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8978_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8978 = { + .probe = wm8978_probe, + .remove = wm8978_remove, + .suspend = wm8978_suspend, + .resume = wm8978_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8978); + +/* + * These registers contain an "update" bit - bit 8. This means, for example, + * that one can write new DAC digital volume for both channels, but only when + * the update bit is set, will also the volume be updated - simultaneously for + * both channels. + */ +static const int update_reg[] = { + WM8978_LEFT_DAC_DIGITAL_VOLUME, + WM8978_RIGHT_DAC_DIGITAL_VOLUME, + WM8978_LEFT_ADC_DIGITAL_VOLUME, + WM8978_RIGHT_ADC_DIGITAL_VOLUME, + WM8978_LEFT_INP_PGA_CONTROL, + WM8978_RIGHT_INP_PGA_CONTROL, + WM8978_LOUT1_HP_CONTROL, + WM8978_ROUT1_HP_CONTROL, + WM8978_LOUT2_SPK_CONTROL, + WM8978_ROUT2_SPK_CONTROL, +}; + +static __devinit int wm8978_register(struct wm8978_priv *wm8978) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8978->codec; + + if (wm8978_codec) { + dev_err(codec->dev, "Another WM8978 is registered\n"); + return -EINVAL; + } + + /* + * Set default system clock to PLL, it is more precise, this is also the + * default hardware setting + */ + wm8978->sysclk = WM8978_PLL; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8978; + codec->name = "WM8978"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8978_set_bias_level; + codec->dai = &wm8978_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8978_CACHEREGNUM; + codec->reg_cache = &wm8978->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8978_reg, sizeof(wm8978_reg)); + + /* + * Set the update bit in all registers, that have one. This way all + * writes to those registers will also cause the update bit to be + * written. + */ + for (i = 0; i < ARRAY_SIZE(update_reg); i++) + ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + + /* Reset the codec */ + ret = snd_soc_write(codec, WM8978_RESET, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8978_dai.dev = codec->dev; + + wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8978_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8978_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8978); + return ret; +} + +static __devexit void wm8978_unregister(struct wm8978_priv *wm8978) +{ + wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8978_dai); + snd_soc_unregister_codec(&wm8978->codec); + kfree(wm8978); + wm8978_codec = NULL; +} + +static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8978_priv *wm8978; + struct snd_soc_codec *codec; + + wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + if (wm8978 == NULL) + return -ENOMEM; + + codec = &wm8978->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8978); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8978_register(wm8978); +} + +static __devexit int wm8978_i2c_remove(struct i2c_client *client) +{ + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + wm8978_unregister(wm8978); + return 0; +} + +static const struct i2c_device_id wm8978_i2c_id[] = { + { "wm8978", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); + +static struct i2c_driver wm8978_i2c_driver = { + .driver = { + .name = "WM8978", + .owner = THIS_MODULE, + }, + .probe = wm8978_i2c_probe, + .remove = __devexit_p(wm8978_i2c_remove), + .id_table = wm8978_i2c_id, +}; + +static int __init wm8978_modinit(void) +{ + return i2c_add_driver(&wm8978_i2c_driver); +} +module_init(wm8978_modinit); + +static void __exit wm8978_exit(void) +{ + i2c_del_driver(&wm8978_i2c_driver); +} +module_exit(wm8978_exit); + +MODULE_DESCRIPTION("ASoC WM8978 codec driver"); +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h new file mode 100644 index 0000000..b58f0bf --- /dev/null +++ b/sound/soc/codecs/wm8978.h @@ -0,0 +1,89 @@ +/* + * wm8978.h -- codec driver for WM8978 + * + * Copyright 2009 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __WM8978_H__ +#define __WM8978_H__ + +/* + * Register values. + */ +#define WM8978_RESET 0x00 +#define WM8978_POWER_MANAGEMENT_1 0x01 +#define WM8978_POWER_MANAGEMENT_2 0x02 +#define WM8978_POWER_MANAGEMENT_3 0x03 +#define WM8978_AUDIO_INTERFACE 0x04 +#define WM8978_COMPANDING_CONTROL 0x05 +#define WM8978_CLOCKING 0x06 +#define WM8978_ADDITIONAL_CONTROL 0x07 +#define WM8978_GPIO_CONTROL 0x08 +#define WM8978_JACK_DETECT_CONTROL_1 0x09 +#define WM8978_DAC_CONTROL 0x0A +#define WM8978_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8978_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8978_JACK_DETECT_CONTROL_2 0x0D +#define WM8978_ADC_CONTROL 0x0E +#define WM8978_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8978_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8978_EQ1 0x12 +#define WM8978_EQ2 0x13 +#define WM8978_EQ3 0x14 +#define WM8978_EQ4 0x15 +#define WM8978_EQ5 0x16 +#define WM8978_DAC_LIMITER_1 0x18 +#define WM8978_DAC_LIMITER_2 0x19 +#define WM8978_NOTCH_FILTER_1 0x1b +#define WM8978_NOTCH_FILTER_2 0x1c +#define WM8978_NOTCH_FILTER_3 0x1d +#define WM8978_NOTCH_FILTER_4 0x1e +#define WM8978_ALC_CONTROL_1 0x20 +#define WM8978_ALC_CONTROL_2 0x21 +#define WM8978_ALC_CONTROL_3 0x22 +#define WM8978_NOISE_GATE 0x23 +#define WM8978_PLL_N 0x24 +#define WM8978_PLL_K1 0x25 +#define WM8978_PLL_K2 0x26 +#define WM8978_PLL_K3 0x27 +#define WM8978_3D_CONTROL 0x29 +#define WM8978_BEEP_CONTROL 0x2b +#define WM8978_INPUT_CONTROL 0x2c +#define WM8978_LEFT_INP_PGA_CONTROL 0x2d +#define WM8978_RIGHT_INP_PGA_CONTROL 0x2e +#define WM8978_LEFT_ADC_BOOST_CONTROL 0x2f +#define WM8978_RIGHT_ADC_BOOST_CONTROL 0x30 +#define WM8978_OUTPUT_CONTROL 0x31 +#define WM8978_LEFT_MIXER_CONTROL 0x32 +#define WM8978_RIGHT_MIXER_CONTROL 0x33 +#define WM8978_LOUT1_HP_CONTROL 0x34 +#define WM8978_ROUT1_HP_CONTROL 0x35 +#define WM8978_LOUT2_SPK_CONTROL 0x36 +#define WM8978_ROUT2_SPK_CONTROL 0x37 +#define WM8978_OUT3_MIXER_CONTROL 0x38 +#define WM8978_OUT4_MIXER_CONTROL 0x39 + +#define WM8978_CACHEREGNUM 58 + +/* Clock divider Id's */ +enum wm8978_clk_id { + WM8978_OPCLKRATE, + WM8978_MCLKDIV, + WM8978_ADCCLK, + WM8978_DACCLK, + WM8978_BCLKDIV, +}; + +enum wm8978_sysclk_src { + WM8978_PLL, + WM8978_MCLK +}; + +extern struct snd_soc_dai wm8978_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8978; + +#endif /* __WM8978_H__ */

2 1

[alsa-devel] Duplicate wake-ups in pcm_lib.c
by pl bossart 27 Jan '10

27 Jan '10

Thanks to Takashi's advice, I managed to find out the reason why I was seeing null events returned by poll(). This could explain why PulseAudio doesn't seem to sleep much. It turns out that we have two calls to wakeup() in pcm_lib.c, and a nice race condition it seems. See the log below. A wake-up is generated during the period interrupt, and a second wake-up is generated during the write loop, after the application was awaken but just before the pointers are updated. This second wake-up shouldn't exist, since the write loop actually fills the ring buffer. By the time the second wake-up is actually handled, there's really no space left in the buffer and a null event is generated; it'll wake-up the application a second time for nothing. Maybe we should move the call to snd_pcm_update_hw_ptr() after the transfer took place? Can anyone apply the attached patches (only adds printks) and confirm this issue on their platforms? That would be very useful. Happy holidays everyone - Pierre alsa-lib/test/pcm --verbose -Dhw -c2 -r48000 -f440 --method write_and_poll [ 371.369865] snd_intel8x0_interrupt [ 371.369876] snd_intel8x0_update [ 371.369884] snd_pcm_period_elapsed [ 371.369890] snd_pcm_update_hw_ptr_interrupt [ 371.369900] snd_pcm_update_hw_ptr_post [ 371.369909] wakeup in snd_pcm_update_hw_ptr_post sound/core/pcm_lib.c: 214, avail 8194, avail_min 8192 [ 371.369935] ALSA: poll event POLLOUT|POLLWRNORM, avail 8194 avail_min 8192 [ 371.375449] snd_pcm_lib_write1: in [ 371.375457] snd_pcm_lib_write1 loop [ 371.375462] snd_pcm_update_hw_ptr [ 371.375472] snd_pcm_update_hw_ptr_post [ 371.375481] wakeup in snd_pcm_update_hw_ptr_post sound/core/pcm_lib.c: 214, avail 8462, avail_min 8192 >>Between these two points, the appl_ptr is updated, and there's no space left in the buffer. The wake-up is generated too early. [ 371.375514] snd_pcm_lib_write1: out [ 371.375524] ALSA: null poll event, avail 270 avail_min 8192 [ 371.540532] snd_intel8x0_interrupt [ 371.540542] snd_intel8x0_update [ 371.540551] snd_pcm_period_elapsed [ 371.540556] snd_pcm_update_hw_ptr_interrupt [ 371.540566] snd_pcm_update_hw_ptr_post [ 371.540575] wakeup in snd_pcm_update_hw_ptr_post sound/core/pcm_lib.c: 214, avail 8194, avail_min 8192 [ 371.540601] ALSA: poll event POLLOUT|POLLWRNORM, avail 8194 avail_min 8192 [ 371.543734] snd_pcm_lib_write1: in [ 371.543741] snd_pcm_lib_write1 loop [ 371.543746] snd_pcm_update_hw_ptr [ 371.543755] snd_pcm_update_hw_ptr_post [ 371.543764] wakeup in snd_pcm_update_hw_ptr_post sound/core/pcm_lib.c: 214, avail 8347, avail_min 8192 [ 371.543797] snd_pcm_lib_write1: out [ 371.543807] ALSA: null poll event, avail 155 avail_min 8192

4 24

[alsa-devel] [PATCHv2 08/10] ARM: SMDK2443: Enable AC97 device
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Add AC97 controller to platform devices and initialize the AC97 gpios. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/mach-s3c2443/mach-smdk2443.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c2443/mach-smdk2443.c b/arch/arm/mach-s3c2443/mach-smdk2443.c index 039a462..5e7d4ab 100644 --- a/arch/arm/mach-s3c2443/mach-smdk2443.c +++ b/arch/arm/mach-s3c2443/mach-smdk2443.c @@ -106,6 +106,9 @@ static struct platform_device *smdk2443_devices[] __initdata = { &s3c_device_wdt, &s3c_device_i2c0, &s3c_device_hsmmc0, +#ifdef CONFIG_SND_SOC_SMDK2443_WM9710 + &s3c_device_ac97, +#endif }; static void __init smdk2443_map_io(void) @@ -118,6 +121,11 @@ static void __init smdk2443_map_io(void) static void __init smdk2443_machine_init(void) { s3c_i2c0_set_platdata(NULL); + +#ifdef CONFIG_SND_SOC_SMDK2443_WM9710 + s3c24xx_ac97_setup_gpio(S3C24XX_AC97_GPE0); +#endif + platform_add_devices(smdk2443_devices, ARRAY_SIZE(smdk2443_devices)); smdk_machine_init(); } -- 1.6.2.5

2 1

[alsa-devel] [PATCH 1/2] ad1836: use soc-cache framework for codec registers access
by Barry Song 27 Jan '10

27 Jan '10

Signed-off-by: Barry Song <Barry.Song(a)analog.com> --- sound/soc/codecs/ad1836.c | 92 +++++++++++--------------------------------- sound/soc/soc-cache.c | 67 ++++++++++++++++++++++++++++++++ 2 files changed, 90 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 83add2f..3c80137 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -171,58 +171,6 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, return 0; } - -/* - * interface to read/write ad1836 register - */ -#define AD1836_SPI_REG_SHFT 12 -#define AD1836_SPI_READ (1 << 11) -#define AD1836_SPI_VAL_MSK 0x3FF - -/* - * write to the ad1836 register space - */ - -static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *reg_cache = codec->reg_cache; - int ret = 0; - - if (value != reg_cache[reg]) { - unsigned short buf; - struct spi_transfer t = { - .tx_buf = &buf, - .len = 2, - }; - struct spi_message m; - - buf = (reg << AD1836_SPI_REG_SHFT) | - (value & AD1836_SPI_VAL_MSK); - spi_message_init(&m); - spi_message_add_tail(&t, &m); - ret = spi_sync(codec->control_data, &m); - if (ret == 0) - reg_cache[reg] = value; - } - - return ret; -} - -/* - * read from the ad1836 register space cache - */ -static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *reg_cache = codec->reg_cache; - - if (reg >= codec->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - #ifdef CONFIG_PM static int ad1836_soc_suspend(struct platform_device *pdev, pm_message_t state) @@ -231,10 +179,10 @@ static int ad1836_soc_suspend(struct platform_device *pdev, struct snd_soc_codec *codec = socdev->card->codec; /* reset clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } static int ad1836_soc_resume(struct platform_device *pdev) @@ -243,10 +191,10 @@ static int ad1836_soc_resume(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->card->codec; /* restore clock control mode */ - u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2); adc_ctrl2 |= AD1836_ADC_AUX; - return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); + return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2); } #else #define ad1836_soc_suspend NULL @@ -336,32 +284,38 @@ static int ad1836_register(struct ad1836_priv *ad1836) codec->owner = THIS_MODULE; codec->dai = &ad1836_dai; codec->num_dai = 1; - codec->write = ad1836_write_reg; - codec->read = ad1836_read_reg_cache; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ad1836_dai.dev = codec->dev; ad1836_codec = codec; + ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad1836); + return ret; + } + /* default setting for ad1836 */ /* de-emphasis: 48kHz, power-on dac */ - codec->write(codec, AD1836_DAC_CTRL1, 0x300); + snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300); /* unmute dac channels */ - codec->write(codec, AD1836_DAC_CTRL2, 0x0); + snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0); /* high-pass filter enable, power-on adc */ - codec->write(codec, AD1836_ADC_CTRL1, 0x100); + snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100); /* unmute adc channles, adc aux mode */ - codec->write(codec, AD1836_ADC_CTRL2, 0x180); + snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180); /* left/right diff:PGA/MUX */ - codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); /* volume */ - codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); - codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF); + snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF); ret = snd_soc_register_codec(codec); if (ret != 0) { diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 02c2357..cde7b63 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -15,6 +15,68 @@ #include <linux/spi/spi.h> #include <sound/soc.h> +static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 4) | ((value >> 8) & 0x000f); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_4_12_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[1]; + msg[1] = data[0]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_4_12_spi_write NULL +#endif + static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -180,6 +242,11 @@ static struct { unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { + .addr_bits = 4, .data_bits = 12, + .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, + .spi_write = snd_soc_4_12_spi_write, + }, + { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, .spi_write = snd_soc_7_9_spi_write, -- 1.5.6.3

2 2

[alsa-devel] [PATCHv2 03/10] ARM: S3C24XX: Update AC97 platform resources
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Modify the AC97 platform backend so that the controller can be run by the s3c-ac97.c Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c/include/plat/audio.h | 4 +++ arch/arm/plat-s3c24xx/devs.c | 45 ++++++++++++++++++++++++++++++- 2 files changed, 47 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h index e07c941..ac56821 100644 --- a/arch/arm/plat-s3c/include/plat/audio.h +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -16,6 +16,10 @@ #define S3C64XX_AC97_GPE 1 extern void s3c64xx_ac97_setup_gpio(int); +#define S3C24XX_AC97_GPE0 0 /* gpe0-4 as AC97 */ +#define S3C24XX_AC97_GPE5 0 /* gpe5-9 as AC97 */ +extern void s3c24xx_ac97_setup_gpio(int); + /** * struct s3c_audio_pdata - common platform data for audio device drivers * @cfg_gpio: Callback function to setup mux'ed pins in I2S/PCM/AC97 mode diff --git a/arch/arm/plat-s3c24xx/devs.c b/arch/arm/plat-s3c24xx/devs.c index 986d4e5..3805317 100644 --- a/arch/arm/plat-s3c24xx/devs.c +++ b/arch/arm/plat-s3c24xx/devs.c @@ -20,6 +20,7 @@ #include <linux/serial_core.h> #include <linux/platform_device.h> #include <linux/io.h> +#include <linux/gpio.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -34,9 +35,11 @@ #include <plat/udc.h> #include <plat/mci.h> +#include <plat/audio.h> #include <plat/devs.h> #include <plat/cpu.h> #include <plat/regs-spi.h> +#include <mach/regs-gpio.h> #include <mach/ts.h> @@ -459,6 +462,10 @@ struct platform_device s3c_device_camif = { EXPORT_SYMBOL(s3c_device_camif); +#endif /* CONFIG_CPU_S32440 */ + +#if defined(CONFIG_CPU_S3C2440) || defined(CONFIG_CPU_S3C2443) + /* AC97 */ static struct resource s3c_ac97_resource[] = { @@ -494,17 +501,51 @@ static struct resource s3c_ac97_resource[] = { static u64 s3c_device_ac97_dmamask = 0xffffffffUL; +static int s3c24xx_ac97_cfg_gpe0(struct platform_device *pdev) +{ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO); + + return 0; +} + +static int s3c24xx_ac97_cfg_gpe5(struct platform_device *pdev) +{ + s3c2410_gpio_cfgpin(S3C2410_GPE5, S3C2443_GPE5_AC_BITCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE6, S3C2443_GPE6_AC_SDI); + s3c2410_gpio_cfgpin(S3C2410_GPE7, S3C2443_GPE7_AC_SDO); + s3c2410_gpio_cfgpin(S3C2410_GPE8, S3C2443_GPE8_AC_SYNC); + s3c2410_gpio_cfgpin(S3C2410_GPE9, S3C2443_GPE9_AC_nRESET); + + return 0; +} + +static struct s3c_audio_pdata s3c_ac97_pdata = { + .cfg_gpio = s3c24xx_ac97_cfg_gpe0, +}; + struct platform_device s3c_device_ac97 = { .name = "s3c-ac97", .id = -1, .num_resources = ARRAY_SIZE(s3c_ac97_resource), .resource = s3c_ac97_resource, .dev = { + .platform_data = &s3c_ac97_pdata, .dma_mask = &s3c_device_ac97_dmamask, .coherent_dma_mask = 0xffffffffUL } }; - EXPORT_SYMBOL(s3c_device_ac97); -#endif // CONFIG_CPU_S32440 +void __init s3c24xx_ac97_setup_gpio(int num) +{ + if (num == S3C24XX_AC97_GPE0) + s3c_ac97_pdata.cfg_gpio = s3c24xx_ac97_cfg_gpe0; + else + s3c_ac97_pdata.cfg_gpio = s3c24xx_ac97_cfg_gpe5; +} + +#endif -- 1.6.2.5

2 1

[alsa-devel] [PATCHv2 02/10] ARM: S3C64XX: Add AC97 platform resources
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> This patch defines the platform device and the resources: IRQ, DMA and MEM, needed by the AC97 controller driver. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c/include/plat/audio.h | 8 +++ arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c64xx/dev-audio.c | 78 ++++++++++++++++++++++++++++++++ 3 files changed, 89 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h index f22d23b..e07c941 100644 --- a/arch/arm/plat-s3c/include/plat/audio.h +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -8,6 +8,14 @@ * published by the Free Software Foundation. */ +/* The machine init code calls s3c*_ac97_setup_gpio with + * one of these defines in order to select appropriate bank + * of GPIO for AC97 pins + */ +#define S3C64XX_AC97_GPD 0 +#define S3C64XX_AC97_GPE 1 +extern void s3c64xx_ac97_setup_gpio(int); + /** * struct s3c_audio_pdata - common platform data for audio device drivers * @cfg_gpio: Callback function to setup mux'ed pins in I2S/PCM/AC97 mode diff --git a/arch/arm/plat-s3c/include/plat/devs.h b/arch/arm/plat-s3c/include/plat/devs.h index c6f9b73..65317eb 100644 --- a/arch/arm/plat-s3c/include/plat/devs.h +++ b/arch/arm/plat-s3c/include/plat/devs.h @@ -35,7 +35,10 @@ extern struct platform_device s3c64xx_device_spi1; extern struct platform_device s3c64xx_device_pcm0; extern struct platform_device s3c64xx_device_pcm1; +extern struct platform_device s3c64xx_device_ac97; + extern struct platform_device s3c_device_ts; + extern struct platform_device s3c_device_fb; extern struct platform_device s3c_device_ohci; extern struct platform_device s3c_device_lcd; diff --git a/arch/arm/plat-s3c64xx/dev-audio.c b/arch/arm/plat-s3c64xx/dev-audio.c index f6b7bfb..63088b7 100644 --- a/arch/arm/plat-s3c64xx/dev-audio.c +++ b/arch/arm/plat-s3c64xx/dev-audio.c @@ -11,6 +11,7 @@ #include <linux/kernel.h> #include <linux/string.h> #include <linux/platform_device.h> +#include <linux/dma-mapping.h> #include <mach/irqs.h> #include <mach/map.h> @@ -254,3 +255,80 @@ struct platform_device s3c64xx_device_pcm1 = { }, }; EXPORT_SYMBOL(s3c64xx_device_pcm1); + +/* AC97 Controller platform devices */ + +static int s3c64xx_ac97_cfg_gpd(struct platform_device *pdev) +{ + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_AC97_BITCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_AC97_nRESET); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_AC97_SYNC); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_AC97_SDI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_AC97_SDO); + + return 0; +} + +static int s3c64xx_ac97_cfg_gpe(struct platform_device *pdev) +{ + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_AC97_BITCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_AC97_nRESET); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_AC97_SYNC); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_AC97_SDI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_AC97_SDO); + + return 0; +} + +static struct resource s3c64xx_ac97_resource[] = { + [0] = { + .start = S3C64XX_PA_AC97, + .end = S3C64XX_PA_AC97 + 0x100 - 1, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = DMACH_AC97_PCMOUT, + .end = DMACH_AC97_PCMOUT, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMACH_AC97_PCMIN, + .end = DMACH_AC97_PCMIN, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DMACH_AC97_MICIN, + .end = DMACH_AC97_MICIN, + .flags = IORESOURCE_DMA, + }, + [4] = { + .start = IRQ_AC97, + .end = IRQ_AC97, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct s3c_audio_pdata s3c_ac97_pdata; + +static u64 s3c64xx_ac97_dmamask = DMA_BIT_MASK(32); + +struct platform_device s3c64xx_device_ac97 = { + .name = "s3c-ac97", + .id = -1, + .num_resources = ARRAY_SIZE(s3c64xx_ac97_resource), + .resource = s3c64xx_ac97_resource, + .dev = { + .platform_data = &s3c_ac97_pdata, + .dma_mask = &s3c64xx_ac97_dmamask, + .coherent_dma_mask = DMA_BIT_MASK(32), + }, +}; +EXPORT_SYMBOL(s3c64xx_device_ac97); + +void __init s3c64xx_ac97_setup_gpio(int num) +{ + if (num == S3C64XX_AC97_GPD) + s3c_ac97_pdata.cfg_gpio = s3c64xx_ac97_cfg_gpd; + else + s3c_ac97_pdata.cfg_gpio = s3c64xx_ac97_cfg_gpe; +} -- 1.6.2.5

2 1

[alsa-devel] [PATCH 1/3] ARM: S3C64XX: Add AC97 platform resources
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> This patch defines the platform device and the resources: IRQ, DMA and MEM, needed by the AC97 controller driver. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- arch/arm/plat-s3c/include/plat/audio.h | 8 +++ arch/arm/plat-s3c/include/plat/devs.h | 3 + arch/arm/plat-s3c64xx/dev-audio.c | 78 ++++++++++++++++++++++++++++++++ 3 files changed, 89 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h index f22d23b..e07c941 100644 --- a/arch/arm/plat-s3c/include/plat/audio.h +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -8,6 +8,14 @@ * published by the Free Software Foundation. */ +/* The machine init code calls s3c*_ac97_setup_gpio with + * one of these defines in order to select appropriate bank + * of GPIO for AC97 pins + */ +#define S3C64XX_AC97_GPD 0 +#define S3C64XX_AC97_GPE 1 +extern void s3c64xx_ac97_setup_gpio(int); + /** * struct s3c_audio_pdata - common platform data for audio device drivers * @cfg_gpio: Callback function to setup mux'ed pins in I2S/PCM/AC97 mode diff --git a/arch/arm/plat-s3c/include/plat/devs.h b/arch/arm/plat-s3c/include/plat/devs.h index c6f9b73..65317eb 100644 --- a/arch/arm/plat-s3c/include/plat/devs.h +++ b/arch/arm/plat-s3c/include/plat/devs.h @@ -35,7 +35,10 @@ extern struct platform_device s3c64xx_device_spi1; extern struct platform_device s3c64xx_device_pcm0; extern struct platform_device s3c64xx_device_pcm1; +extern struct platform_device s3c64xx_device_ac97; + extern struct platform_device s3c_device_ts; + extern struct platform_device s3c_device_fb; extern struct platform_device s3c_device_ohci; extern struct platform_device s3c_device_lcd; diff --git a/arch/arm/plat-s3c64xx/dev-audio.c b/arch/arm/plat-s3c64xx/dev-audio.c index f6b7bfb..63088b7 100644 --- a/arch/arm/plat-s3c64xx/dev-audio.c +++ b/arch/arm/plat-s3c64xx/dev-audio.c @@ -11,6 +11,7 @@ #include <linux/kernel.h> #include <linux/string.h> #include <linux/platform_device.h> +#include <linux/dma-mapping.h> #include <mach/irqs.h> #include <mach/map.h> @@ -254,3 +255,80 @@ struct platform_device s3c64xx_device_pcm1 = { }, }; EXPORT_SYMBOL(s3c64xx_device_pcm1); + +/* AC97 Controller platform devices */ + +static int s3c64xx_ac97_cfg_gpd(struct platform_device *pdev) +{ + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_AC97_BITCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_AC97_nRESET); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_AC97_SYNC); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_AC97_SDI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_AC97_SDO); + + return 0; +} + +static int s3c64xx_ac97_cfg_gpe(struct platform_device *pdev) +{ + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_AC97_BITCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_AC97_nRESET); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_AC97_SYNC); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_AC97_SDI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_AC97_SDO); + + return 0; +} + +static struct resource s3c64xx_ac97_resource[] = { + [0] = { + .start = S3C64XX_PA_AC97, + .end = S3C64XX_PA_AC97 + 0x100 - 1, + .flags = IORESOURCE_MEM, + }, + [1] = { + .start = DMACH_AC97_PCMOUT, + .end = DMACH_AC97_PCMOUT, + .flags = IORESOURCE_DMA, + }, + [2] = { + .start = DMACH_AC97_PCMIN, + .end = DMACH_AC97_PCMIN, + .flags = IORESOURCE_DMA, + }, + [3] = { + .start = DMACH_AC97_MICIN, + .end = DMACH_AC97_MICIN, + .flags = IORESOURCE_DMA, + }, + [4] = { + .start = IRQ_AC97, + .end = IRQ_AC97, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct s3c_audio_pdata s3c_ac97_pdata; + +static u64 s3c64xx_ac97_dmamask = DMA_BIT_MASK(32); + +struct platform_device s3c64xx_device_ac97 = { + .name = "s3c-ac97", + .id = -1, + .num_resources = ARRAY_SIZE(s3c64xx_ac97_resource), + .resource = s3c64xx_ac97_resource, + .dev = { + .platform_data = &s3c_ac97_pdata, + .dma_mask = &s3c64xx_ac97_dmamask, + .coherent_dma_mask = DMA_BIT_MASK(32), + }, +}; +EXPORT_SYMBOL(s3c64xx_device_ac97); + +void __init s3c64xx_ac97_setup_gpio(int num) +{ + if (num == S3C64XX_AC97_GPD) + s3c_ac97_pdata.cfg_gpio = s3c64xx_ac97_cfg_gpd; + else + s3c_ac97_pdata.cfg_gpio = s3c64xx_ac97_cfg_gpe; +} -- 1.6.2.5

4 21

[alsa-devel] How to unload a sound module.
by Yiliang Bao 27 Jan '10

27 Jan '10

Hi, I am new to alsa driver development. The first problem I encountered is I could not unload a module just inserted. I always get the following message (using snd-dummy as an example). > sudo rmmod snd-dummy ERROR: Module snd_dummy is in use I checked who is using snd-dummy. Here is what I got and I do not know what applications are using it. Module Size Used by snd_dummy 9600 2 Is there any easy way to unload the sound module? Thanks, Yiliang

2 1

[alsa-devel] [PATCHv2 10/10] ASoC: AC97: S3C2443: Remove unused driver
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Since, we have generic AC97 controller driver and all the machines have moved to that, there is no need for old s3c2443-ac97.c to exist. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 3 - sound/soc/s3c24xx/Makefile | 2 - sound/soc/s3c24xx/s3c2443-ac97.c | 432 -------------------------------------- sound/soc/s3c24xx/s3c24xx-ac97.h | 25 --- 4 files changed, 0 insertions(+), 462 deletions(-) delete mode 100644 sound/soc/s3c24xx/s3c2443-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-ac97.h diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 69d143e..15fe57e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -27,9 +27,6 @@ config SND_S3C64XX_SOC_I2S config SND_S3C_SOC_PCM tristate -config SND_S3C2443_SOC_AC97 - tristate - config SND_S3C_SOC_AC97 tristate select SND_SOC_AC97_BUS diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 1117678..df071a3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,14 +3,12 @@ snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c deleted file mode 100644 index 0191e3a..0000000 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ /dev/null @@ -1,432 +0,0 @@ -/* - * s3c2443-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright (C) 2005, Sean Choi <sh428.choi(a)samsung.com> - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/io.h> -#include <linux/wait.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/clk.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/ac97_codec.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <plat/regs-ac97.h> -#include <mach/regs-gpio.h> -#include <mach/regs-clock.h> -#include <asm/dma.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "s3c24xx-ac97.h" - -struct s3c24xx_ac97_info { - void __iomem *regs; - struct clk *ac97_clk; -}; -static struct s3c24xx_ac97_info s3c24xx_ac97; - -static DECLARE_COMPLETION(ac97_completion); -static u32 codec_ready; -static DEFINE_MUTEX(ac97_mutex); - -static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - stat = readl(s3c24xx_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," - " rep addr = %02x\n", reg, addr); - - mutex_unlock(&ac97_mutex); - - return (unsigned short)data; -} - -static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl; - u32 ac_codec_cmd; - - mutex_lock(&ac97_mutex); - - codec_ready = S3C_AC97_GLBSTAT_CODECREADY; - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - wait_for_completion(&ac97_completion); - - ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&ac97_mutex); - -} - -static void s3c2443_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_WARMRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c2443_ac97_cold_reset(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA | - S3C_AC97_GLBCTRL_PCMINTM_DMA | S3C_AC97_GLBCTRL_MICINTM_DMA; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); -} - -static irqreturn_t s3c2443_ac97_irq(int irq, void *dev_id) -{ - int status; - u32 ac_glbctrl; - - status = readl(s3c24xx_ac97.regs + S3C_AC97_GLBSTAT) & codec_ready; - - if (status) { - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - complete(&ac97_completion); - } - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c2443_ac97_read, - .write = s3c2443_ac97_write, - .warm_reset = s3c2443_ac97_warm_reset, - .reset = s3c2443_ac97_cold_reset, -}; - -static struct s3c2410_dma_client s3c2443_dma_client_out = { - .name = "AC97 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_in = { - .name = "AC97 PCM Stereo in" -}; - -static struct s3c2410_dma_client s3c2443_dma_client_micin = { - .name = "AC97 Mic Mono in" -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { - .client = &s3c2443_dma_client_out, - .channel = DMACH_PCM_OUT, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { - .client = &s3c2443_dma_client_in, - .channel = DMACH_PCM_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { - .client = &s3c2443_dma_client_micin, - .channel = DMACH_MIC_IN, - .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, - .dma_size = 4, -}; - -static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - u32 ac_glbctrl; - - s3c24xx_ac97.regs = ioremap(S3C2440_PA_AC97, 0x100); - if (s3c24xx_ac97.regs == NULL) - return -ENXIO; - - s3c24xx_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (s3c24xx_ac97.ac97_clk == NULL) { - printk(KERN_ERR "s3c2443-ac97 failed to get ac97_clock\n"); - iounmap(s3c24xx_ac97.regs); - return -ENODEV; - } - clk_enable(s3c24xx_ac97.ac97_clk); - - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2443_GPE0_AC_nRESET); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2443_GPE1_AC_SYNC); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2443_GPE2_AC_BITCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2443_GPE3_AC_SDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2443_GPE4_AC_SDO); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_COLDRESET; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = 0; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); - } - return ret; -} - -static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - free_irq(IRQ_S3C244x_AC97, NULL); - clk_disable(s3c24xx_ac97.ac97_clk); - clk_put(s3c24xx_ac97.ac97_clk); - iounmap(s3c24xx_ac97.regs); -} - -static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; - else - cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_in; - - return 0; -} - -static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - break; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - cpu_dai->dma_data = &s3c2443_ac97_mic_mono_in; - - return 0; -} - -static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; - - ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - } - writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -#define s3c2443_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) - -static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c2443_ac97_dai[] = { -{ - .name = "s3c2443-ac97", - .id = 0, - .ac97_control = 1, - .probe = s3c2443_ac97_probe, - .remove = s3c2443_ac97_remove, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_dai_ops, -}, -{ - .name = "pxa2xx-ac97-mic", - .id = 1, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = s3c2443_AC97_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c2443_ac97_mic_dai_ops, -}, -}; -EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int __init s3c2443_ac97_init(void) -{ - return snd_soc_register_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_init(s3c2443_ac97_init); - -static void __exit s3c2443_ac97_exit(void) -{ - snd_soc_unregister_dais(s3c2443_ac97_dai, - ARRAY_SIZE(s3c2443_ac97_dai)); -} -module_exit(s3c2443_ac97_exit); - - -MODULE_AUTHOR("Graeme Gregory"); -MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h deleted file mode 100644 index e96f941..0000000 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * s3c24xx-ac97.c -- ALSA Soc Audio Layer - * - * (c) 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXAC97_H_ -#define S3C24XXAC97_H_ - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -extern struct snd_soc_dai s3c2443_ac97_dai[]; - -#endif /*S3C24XXAC97_H_*/ -- 1.6.2.5

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[alsa-devel] [PATCHv2 09/10] ASoC: AC97: LN2440SBC: Switch to s3c-ac97.c
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 7 +++---- sound/soc/s3c24xx/ln2440sbc_alc650.c | 4 ++-- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 8b62798..69d143e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -29,9 +29,6 @@ config SND_S3C_SOC_PCM config SND_S3C2443_SOC_AC97 tristate - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_BUS config SND_S3C_SOC_AC97 tristate @@ -86,8 +83,10 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 config SND_S3C24XX_SOC_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index d00d359..ffa954f 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -25,7 +25,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card ln2440sbc; @@ -33,7 +33,7 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- 1.6.2.5

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[alsa-devel] [PATCHv2 07/10] ASoC: AC97: SMDK2443: Switch to s3c-ac97.c
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Switch to use s3c-ac97.c AC97 controller driver. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 +++- sound/soc/s3c24xx/smdk2443_wm9710.c | 4 ++-- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index d1c6f93..8b62798 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -75,8 +75,10 @@ config SND_S3C64XX_SOC_WM8580 config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select SND_S3C2443_SOC_AC97 + select S3C2410_DMA + select AC97_BUS select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 help Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 12b783b..3622588 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -21,7 +21,7 @@ #include "../codecs/ac97.h" #include "s3c-dma.h" -#include "s3c24xx-ac97.h" +#include "s3c-ac97.h" static struct snd_soc_card smdk2443; @@ -29,7 +29,7 @@ static struct snd_soc_dai_link smdk2443_dai[] = { { .name = "AC97", .stream_name = "AC97 HiFi", - .cpu_dai = &s3c2443_ac97_dai[0], + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], .codec_dai = &ac97_dai, }, }; -- 1.6.2.5

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[alsa-devel] [PATCHv2 04/10] ASoC: AC97: S3C: Add controller driver
by jassisinghbrar＠gmail.com 27 Jan '10

27 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Add the AC97 controller driver for Samsung SoCs that have one. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 6 +- sound/soc/s3c24xx/Makefile | 3 +- sound/soc/s3c24xx/s3c-ac97.c | 518 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-ac97.h | 23 ++ 4 files changed, 548 insertions(+), 2 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-ac97.c create mode 100644 sound/soc/s3c24xx/s3c-ac97.h diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b489f1a..ad3690e 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -32,7 +32,11 @@ config SND_S3C2443_SOC_AC97 select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_BUS - + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + config SND_S3C24XX_SOC_NEO1973_WM8753 tristate "SoC I2S Audio support for NEO1973 - WM8753" depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b744657..b7411bd 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,12 +4,14 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o +snd-soc-s3c-ac97-objs := s3c-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o @@ -37,4 +39,3 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o - diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c new file mode 100644 index 0000000..ee8ed9d --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -0,0 +1,518 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/soc.h> + +#include <plat/regs-ac97.h> +#include <mach/dma.h> +#include <plat/audio.h> + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_dai->dma_data = &s3c_ac97_pcm_out; + else + cpu_dai->dma_data = &s3c_ac97_pcm_in; + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + cpu_dai->dma_data = &s3c_ac97_mic_in; + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h new file mode 100644 index 0000000..2781983 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ -- 1.6.2.5

1 0

[alsa-devel] Problem with audio card
by Angelo D. 26 Jan '10

26 Jan '10

Hi, i have a problem with my audio card. I hear sound from headphones but i can't hear sound without headphones.This is the result of the esecution alsa-info.sh: http://www.alsa-project.org/db/?f=4a72836b153be44d5a3b84bce66638fe41afadb7 This is the link to my related post on ubuntuforum: http://ubuntuforums.org/showthread.php?p=8707917#post8707917 Angelo Di Gennaro

1 0

[alsa-devel] Problem with audio card
by Angelo D. 26 Jan '10

26 Jan '10

Hi, i have a problem with my audio card. I hear sound from headphones but i can't hear sound without headphones.This is the result of the esecution alsa-info.sh: http://www.alsa-project.org/db/?f=4a72836b153be44d5a3b84bce66638fe41afadb7 This is the link to my related post on ubuntuforum: http://ubuntuforums.org/showthread.php?p=8707917#post8707917 Angelo Di Gennaro Casa dei Diritti Sociali: Perchè no? Responsabile di Zona - ArcipelagoSCEC Caserta 3881736622

1 0

[alsa-devel] Recommendations for 4-channel USB-connected audio interface
by Steve Strobel 26 Jan '10

26 Jan '10

I am in need of an ALSA-compatible audio interface that will connect to a PC via USB and record/play four or more audio channels simultaneously. Any recommendations (or warnings) about specific products or chipsets would be greatly appreciated. High resolution and sample rates aren't necessary. The rest of the story is that my company is working on a project using an embedded x86 computer to interface with four independent mono audio devices. I will be responsible for the audio processing and want to make sure that we use an audio chipset for which an ALSA driver is already available. I would also prefer that it interface via USB so I can easily test on my development PC, then move it to the embedded system. We will eventually put the audio chipset on a custom PCB, so it is the audio chipset and driver that are my primary concerns, but a readily-available audio interface using the chipset would serve as a nice prototype. Thanks for any suggestions. Steve --- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

1 0

[alsa-devel] [PATCHv2 5/6] ASoC: OMAP4: Add support for McPDM
by Candelaria Villareal, Jorge 26 Jan '10

26 Jan '10

McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 474 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 152 +++++++++++++++ 4 files changed, 631 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..b96e9d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..bf8b388 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..2bc5acd --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,474 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/irq.h> + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset(int links, int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (links & MCPDM_UPLINK) { + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + } + + if (links & MCPDM_DOWNLINK) { + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + } + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_set_uplink(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_set_downlink(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_clr_uplink(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_clr_downlink(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_DOWNLINK, 1); + omap_mcpdm_set_downlink(mcpdm_irq->downlink); + omap_mcpdm_reset(MCPDM_DOWNLINK, 0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_DOWNLINK, 1); + omap_mcpdm_set_downlink(mcpdm_irq->downlink); + omap_mcpdm_reset(MCPDM_DOWNLINK, 0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_UPLINK, 1); + omap_mcpdm_set_uplink(mcpdm_irq->uplink); + omap_mcpdm_reset(MCPDM_UPLINK, 0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset(MCPDM_UPLINK, 1); + omap_mcpdm_set_uplink(mcpdm_irq->uplink); + omap_mcpdm_reset(MCPDM_UPLINK, 0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + return -EBUSY; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + return ret; + } + + return 0; +} + +void omap_mcpdm_free(void) +{ + clk_disable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* Enable/disable offset cancellation for downlink channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* Enable/disable offset cancellation for downlink channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto exit; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_ioremap; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_ioremap: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_disable(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + int ret; + + ret = platform_driver_register(&omap_mcpdm_driver); + if (ret) + goto error; + return 0; + +error: + return ret; +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..e70f51c --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,152 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_reg_dump(void); +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_set_uplink(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_set_downlink(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_clr_uplink(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_clr_downlink(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- 1.6.0.4

4 6

[alsa-devel] [PATCHv2 6/6] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 26 Jan '10

26 Jan '10

From: Misael Lopez Cruz <x0052729(a)ti.com> McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/omap-mcpdm.c | 250 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 3 files changed, 280 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index bf8b388..d6382fa 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := mcpdm.o +snd-soc-omap-mcpdm-objs := mcpdm.o omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..180cca6 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,250 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/control.h> +#include <mach/dma.h> +#include <mach/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio downlink", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio uplink", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream) + /* up to 2 channels for uplink */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream) { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_set_uplink(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_set_downlink(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int err; + + if (substream->stream) + err = omap_mcpdm_clr_uplink(&mcpdm_links[substream->stream]); + else + err = omap_mcpdm_clr_downlink(&mcpdm_links[substream->stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

4 6

[alsa-devel] substream usage
by Singaravelan Nallasellan 26 Jan '10

26 Jan '10

Hi, In my system, I have three microphones to capture. Can I represent each of channels of a single capture stream as substream? How will be able include or exclude a substream to the stream? Thanks Sing

1 0

[alsa-devel] [PATCH 0/4] ALSA: SH: add ASoC driver for SIU audio engine, an audio codec and platform support
by Guennadi Liakhovetski 26 Jan '10

26 Jan '10

Hi This patch series adds support for the Sound Interface Unit (SIU), found on several SuperH SoCs, including sh7722, sh7343, sh7354, sh7367. The driver supports playback and capture at various bitrates, mono and stereo, so far only tested with an I2S codec, support for the PCM (SND_SOC_DAIFMT_LEFT_J) is also included. The hardware also supports S/PDIF, but including support for it would require more work and testing on a suitable hardware platform. The SIU engine includes a DSP, that has to be programmed separately. The driver uses the standard request_firmware() API to load the program and data. This patch series also requires the earlier [PATCH] sh: support SIU sourcing from external clock on sh7722 (http://www.spinics.net/lists/linux-sh/msg04075.html) patch for proper functionality. Comments welcome. Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

3 24

[alsa-devel] [PATCH] ASoC: DaVinci: Fix stream restart error
by Chaithrika U S 26 Jan '10

26 Jan '10

Sometimes after a suspend-resume cycle, the ALSA application restarts the stream when resume fails and McASP fails to work as the clock is not enabled. This patch corrects this bug. Testes on TI DA850/OMAP-L138 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-mcasp.c | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a613bbb..ab6518d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!dev->clk_active) { clk_enable(dev->clk); dev->clk_active = 1; } - /* Fall through */ - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_mcasp_start(dev, substream->stream); break; -- 1.5.6

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Updates to EVM hardware params callback
by Chaithrika U S 26 Jan '10

26 Jan '10

For S/PDIF transciever codec to work on TI DM6467 EVM, the codec ops pointer should be checked for in the params callback. This dummy codec has no set_fmt or set_sysclk implementations. Tested on TI DM6467 EVM. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> Tested-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-evm.c | 17 +++++++++++------ 1 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..1760354 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -64,9 +64,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); - if (ret < 0) - return ret; + if (codec_dai->ops->set_fmt) { + ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); + if (ret < 0) + return ret; + } /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); @@ -74,9 +76,12 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return ret; /* set the codec system clock */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; + if (codec_dai->ops->set_sysclk) { + ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + } return 0; } -- 1.5.6

2 1

[alsa-devel] [PATCH]Add support for more the 8 streams
by Wei Ni 26 Jan '10

26 Jan '10

Hi, Takashi When I develop the hdmi audio driver on my new chipset, I found a bug in the hda_intel.c It use azx_readb/azx_writeb to read/write SIE, so it just support 8 streams. The new HAD controller has more than 8 streams, so I think it should use azx_readl/azx_writel. I attached the patch file, please check it. Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

2 1

[alsa-devel] [PATCH 1/5] MFD: DaVinci Voice Codec
by miguel.aguilar＠ridgerun.com 26 Jan '10

26 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 ++++++++++++++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++++++++++ 4 files changed, 320 insertions(+), 0 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..6e93c0b 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,10 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + tristate + select MFD_CORE + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..9886aa8 --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,189 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + struct mfd_cell *cell = NULL; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + davinci_vc->pdev = pdev; + + /* Voice codec interface client */ + cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; + cell->name = "davinci_vcif"; + cell->driver_data = davinci_vc; + + /* Voice codec CQ93VC client */ + cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; + cell->name = "cq93vc"; + cell->driver_data = davinci_vc; + + ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, + DAVINCI_VC_CELLS, NULL, 0); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client devices\n"); + goto fail4; + } + + return 0; + +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + mfd_remove_devices(&pdev->dev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..0ab6132 --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,126 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/mfd/core.h> + +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +#define DAVINCI_VC_CELLS 2 + +enum davinci_vc_cells { + DAVINCI_VC_VCIF_CELL, + DAVINCI_VC_CQ93VC_CELL, +}; + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct platform_device *pdev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* MFD cells */ + struct mfd_cell cells[DAVINCI_VC_CELLS]; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif -- 1.6.0.4

3 2

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 26 Jan '10

26 Jan '10

The following changes since commit 821dd91ec7838e1313d783384ea9ce43510d4013: Mark Brown (1): ASoC: Use BIAS_OFF when idle for wm_hubs devices are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 The changelog and diffstat look enormous due to the merge up of 2.6.33-rc5 into the fixes branch. Abhijeet Kolekar (1): iwlwifi: disable tx on beacon update notification Abhijit Pagare (1): ARM: OMAP3: PM: Fix the Invalid CM_CLKSTCTRL reg access. Adam Jackson (3): Input: evdev - be less aggressive about sending SIGIO notifies drm/edid: Skip empty CVT codepoints drm/edid: Fix CVT width/height decode Adrian Hunter (2): mmc_block: fix queue cleanup mmc: allow for MMC v4.4 Al Viro (13): mn10300: fix several bogus includes on abs2305 fix autofs/afs/etc. magic mountpoint breakage Kill 9p readlink() hppfs: handle ->put_link() Unrot uml mconsole a bit Fix ACC_MODE() for real Fix the -ESTALE handling in do_filp_open() Fix configfs leak may_umount() needs namespace_sem attach_recursive_mnt() needs to hold vfsmount_lock over set_mnt_shared() mnt_flags fixes in do_remount() CIFS shouldn't make mountpoints shrinkable do_add_mount() should sanitize mnt_flags Alan Carvalho de Assis (1): mx27: mxt_td60: Remove not used UART pins Alan Cox (6): i2o: propogate the BKL down into the ioctl method tosh: Use non bkl ioctl x86: mce.h: Fix warning in header checks nozomi: quick fix for the close/close bug serial: Fix crash if the minimum rate of the device is > 9600 baud Staging: et131x: Fix 2.6.33rc1 regression in et131x Alan Jenkins (1): eeepc-laptop: disable cpu speed control on EeePC 701 Alan Stern (4): USB: fix bitmask merge error USB: EHCI: fix handling of unusual interrupt intervals USB: EHCI & UHCI: fix race between root-hub suspend and port resume USB: add missing delay during remote wakeup Alberto Panizzo (1): mfd: Unlock mc13783 before subsystems initialisation, at probe time. Albin Tonnerre (4): lib: add support for LZO-compressed kernels arm: add support for LZO-compressed kernels x86: add support for LZO-compressed kernels Add LZO compression support for initramfs and old-style initrd Alex Chiang (1): ACPI: processor: restrict early _PDC to opt-in platforms Alex Deucher (13): drm/radeon/kms: fallback to default connector table drm/radeon/kms: add primary dac adj values table drm/radeon/kms: add missing breaks in i2c and ss lookups drm/radeon/kms: fix typo in atom connector type handling drm/radeon/kms: whitespace changes to ObjectID.h drm/radeon/kms: pull in the latest upstream ObjectID.h changes drm: Add eDP connector type drm/radeon/kms: add support for eDP (embedded DisplayPort) drm/radeon/kms: detect sideport memory on IGP chips drm/radeon/kms: add additional safe regs for r4xx/rs6xx and r5xx drm/radeon/kms: fix up LVDS handling on macs (v2) drm/radeon/kms/r6xx+: make irq handler less verbose drm/radeon/kms/rv100: reject modes > 135 Mhz on DVI (v2) Alex Murray (1): ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support Alexander Beregalov (3): pcmcia: ncmlan_cs: remove odd bracket drm/radeon: mkregtable.c: close a file before exit USB: FHCI: avoid NULL pointer dereference Alexander Duyck (5): igb: do not force pcs link when in KX mode igb: do not force retry count to 1 on 82580 phy igb: correctly offset 82575 flow control watermarks by 16 bytes igb: check both function bits in status register in wol exception ixgbe: Do not attempt to perform interrupts in netpoll when down Alexander Graf (1): KVM: powerpc: Fix mtsrin in book3s_64 mmu Alexander Shishkin (1): omap2/3: make serial_in_override() address the right uart port Alexey Starikovskiy (3): ACPI: EC: Accelerate query execution ACPI: SBS: Move SBS HC callback to faster Notify queue ACPI: EC: Add wait for irq storm Amit Kumar Salecha (4): netxen: fix tx ring memory leak netxen: fix smatch warning netxen: fix set mac addr netxen: update version to 4.0.72 Ananth N Mavinakayanahalli (1): Revert "x86, apic: Use logical flat on intel with <= 8 logical cpus" Andi Kleen (6): kernel/signal.c: fix kernel information leak with print-fatal-signals=1 kfifo: use void * pointers for user buffers kfifo: sanitize *_user error handling kfifo: add kfifo_out_peek kfifo: add kfifo_initialized kfifo: document everywhere that size has to be power of two Andrea Arcangeli (1): mm: hugetlb: fix clear_huge_page() Andreas Fenkart (1): mm: make totalhigh_pages unsigned long Andrew Lutomirski (1): drm/i915: Fix RC6 suspend/resume Andrew Morton (6): aoe: switch to the new bio_flush_dcache_pages() interface ext4: fix unsigned long long printk warning in super.c jbd2: don't use __GFP_NOFAIL in journal_init_common() net/sctp/socket.c: squish warning drm/i915: fix unused var percpu: avoid calling __pcpu_ptr_to_addr(NULL) Andrew Vasquez (1): [SCSI] qla2xxx: Extend base EEH support in qla2xxx. Andrey Borzenkov (1): orinoco: fix GFP_KERNEL in orinoco_set_key with interrupts disabled Aneesh Kumar K.V (1): ext4: Ensure zeroout blocks have no dirty metadata Anil Ravindranath (1): [SCSI] pmcraid: fix to avoid twice scsi_dma_unmap for a command Anirban Chakraborty (1): [SCSI] qla2xxx: Fix for a multiqueue bug in CPU affinity mode Anisse Astier (1): hp-wmi: remove double free caused by merge conflict Anna Lemehova (1): mmc_block: add dev_t initialization check Anton Blanchard (5): [IA64] cpumask_of_node() should handle -1 as a node MIPS: cpumask_of_node() should handle -1 as a node alpha: cpumask_of_node() should handle -1 as a node powerpc/pseries: Fix xics interrupt affinity powerpc: cpumask_of_node() should handle -1 as a node Anton Vorontsov (8): ucc_geth: Fix empty TX queue processing ucc_geth: Fix netdev watchdog triggering on link changes ucc_geth: Don't needlessly change MAC settings in adjust_link() phylib: Fix deadlock on resume phylib: Properly reinitialize PHYs after hibernation ucc_geth: Fix netdev watchdog triggering on suspend fsl_pq_mdio: Fix iomem unmapping for non-eTSEC2.0 controllers powerpc/swsusp_32: Fix TLB invalidation Arjan van de Ven (2): ipvs: Add boundary check on ioctl arguments perf timechart: Use tid not pid for COMM change Arnaldo Carvalho de Melo (5): perf diff: Fix usage array, it must end with a NULL entry perf record: We should fork only if a program was specified to run perf tools: Add missing header files to LIB_H Makefile variable perf tools: Move QUIET_STDERR def to before first use perf tools: Check if /dev/null can be used as the -o gcc argument Arnaud Patard (2): ARM: S3C24XX: touchscreen device definition ARM: S3C24XX: touchscreen device definition Atsushi Nemoto (2): MIPS: TXx9: Cleanup builtin-cmdline processing compat_ioctl: Supress "unknown cmd" message on serial /dev/console Avi Kivity (1): core, x86: make LIST_POISON less deadly Bahadir Balban (1): ARM: 5858/1: Remove unused vma_vm_flags macro from v7wbi_flush_user_tlb_range Barry Song (1): ASoC: ad1836: reset and restore clock control mode in suspend/resume entry Baruch Siach (3): mx25: s/NO_PAD_CTL/NO_PAD_CTRL/ mx25: add support for FEC on i.MX25 mx25: pdk: add platform code for FEC support Ben Dooks (7): ARM: mach-osiris: add NAND_SCAN_SILENT_NODEV to optional devices ARM: mach-bast: add NAND_SCAN_SILENT_NODEV to optional devices ARM: S3C: Fix NAND device registration by s3c_nand_set_platdata(). ARM: Merge next-simtec ARM: Merge s3c24xx-updates2 ARM: S3C64XX: Fix possible clock look in EPLL and MPLL clock chains ARM: MINI2440: Fixup __initdata usage Ben Hutchings (9): sfc: Include XGXS in XMAC link status check except in XGMII loopback sfc: QT2025C: Add error message for suspected bad SFP+ cables sfc: Disable TX descriptor prefetch watchdog via-velocity: Give RX descriptors to the NIC later on open or MTU change modules: Skip empty sections when exporting section notes dmfe/tulip: Let dmfe handle DM910x except for SPARC on-board chips Documentation/3c509: document ethtool support sfc: Fix polling for slow MCDI operations sfc: Fix conditions for MDIO self-test Ben Skeggs (9): drm/nv50: ignore vbios table's claim to the contrary if EDID says >8bpc drm/nouveau: fix handling of fbcon colours in 8bpp drm/nouveau: remove unused nouveau_channel_idle() function drm/nv50: restore correct cache1 get/put address on fifoctx load drm/nouveau: have ttm's fault handler called directly drm/nv50: prevent a possible ctxprog hang drm/nv04: differentiate between nv04/nv05 drm/nouveau: use dma.max rather than pushbuf size for checking GET validity drm/nouveau: initialise DMA tracking parameters earlier Benjamin Herrenschmidt (6): PCI/cardbus: Add a fixup hook and fix powerpc ps3_gelic_wireless: Fix build failure due to missing WEXT_PRIV zlib: Fix build of powerpc boot wrapper serial/pmac_zilog: Workaround problem due to interrupt on closed port Merge commit 'kumar/merge' into merge Merge commit 'gcl/merge' into merge Benoit Papillault (2): ath9k: Last fix for TX software padding. zd1211rw: Fix multicast filtering. Bernard Pidoux F6BVP (1): rose_loopback_timer sets VC number <= ROSE_DEFAULT_MAXVC Bjorn Helgaas (3): agp/hp: fixup hp agp after ACPI changes agp/hp: fail gracefully if we don't find an IOC mn10300: use generic pci_enable_resources() Boaz Harrosh (2): exofs: fix pnfs_osd re-definitions in pre-pnfs trees exofs: simple_write_end does not mark_inode_dirty Bob Copeland (1): ath5k: fix SWI calibration interrupt storm Bruce Allan (7): e1000e: call pci_save_state() after pci_restore_state() e1000e: don't accumulate PHY statistics on PHY read failure e1000e: perform 10/100 adaptive IFS only on parts that support it e1000e: e1000e_enable_tx_pkt_filtering() returns wrong value e1000e: fix and commonize code for setting the receive address registers e1000e: MDIO slow mode should always be done for 82577 e1000e: workaround link issues on busy hub in half duplex on 82577/82578 Bryn M. Reeves (1): [SCSI] megaraid_sas: remove sysfs poll_mode_io world writeable permissions Carlos Corbacho (1): ACPI: WMI: Survive BIOS with duplicate GUIDs Chen Gong (1): acpi_pad: fix error checks Chris Mason (1): Btrfs: deal with NULL acl sent to btrfs_set_acl Chris Wilson (2): drm/i915: Hold struct mutex whilst pinning power context bo. drm/i915: Permit pinning whilst the device is 'suspended' Christian Schuerer-Waldheim (1): HID: add device IDs for new model of Apple Wireless Keyboard Christoph Hellwig (3): nfsd: make sure data is on disk before calling ->fsync xfs: use DECLARE_EVENT_CLASS xfs: fix timestamp handling in xfs_setattr Clemens Ladisch (2): firewire: fix use of multiple AV/C devices, allow multiple FCP listeners hwmon: (k10temp) Blacklist more family 10h processors Colin Tuckley (2): ARM: 5873/1: ARM: Fix the reset logic for ARM RealView boards USB: Fix level of isp1760 Reloading ptd error message Corbin Simpson (1): drm/radeon/kms: Workaround RV410/R420 CP errata (V3) Corentin Chary (3): eeepc-laptop: dmi blacklist to disable pci hotplug code eeepc-laptop: add hotplug_disable parameter eeepc-laptop: disable wireless hotplug for 1201N Cory Maccarrone (4): omap: gpio: Simultaneously requested rising and falling edge omap1: Add 7xx clocks and pin muxes for SPI OMAP1 clock: Add missing clocks for OMAP 7xx OMAP1 clock: remove __initdata from struct clk_functions to prevent crash Csaba Henk (1): PCI: Handle case when no pci device can provide cache line size hint Cyril Hrubis (1): [ARM] pxa: fix strange characters in zaurus gpio .desc Cyrill Gorcunov (1): x86: kernel_thread() -- initialize SS to a known state Daisuke HATAYAMA (1): binfmt_elf_fdpic: Fix build breakage introduced by coredump changes. Daisuke Nishimura (1): memcg: ensure list is empty at rmdir Dan Carpenter (10): bond_3ad.c avoid possible null deref wl1271_cmd.c: cleanup char => u8 hamradio: avoid null deref v3 rrunner: fix buffer overflow iwl: off by one bug sound: oss: off by one bug iwlwifi: silence buffer overflow warning V4L/DVB (13955): cx25821: fix double unlock in medusa_video_init() mfd: WM8350 off by one bug ecryptfs: use after free Dan Williams (5): ioat3: fix p-disabled q-continuation async_tx: expand async raid6 test to cover ioatdma corner case ioat2,3: put channel hardware in known state at init Merge branch 'ioat' into fixes md: make recovery started by do_md_run() visible via sync_action Daniel Drake (2): Fix MAC address access in 3c507, ibmlana, pcnet32 and libertas Input: psmouse - fix Synaptics detection when protocol is disabled Daniel Mack (1): Libertas: fix buffer overflow in lbs_get_essid() Daniel T Chen (2): ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2 ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist Daniel Vetter (1): drm/i915: fix order of fence release wrt flushing Darren Jenkins (6): drm/radeon/radeon_connectors.c: add a NULL test before dereference drm/radeon/radeon_fence.c: move a dereference below the NULL test drm/radeon/radeon_device.c: move a dereference below a NULL test gpu/drm/radeon/radeon_irq.c: move a dereference below a NULL test drm/radeon: fix a couple of array index errors ACPI: power_meter: remove double kfree() Dave Airlie (9): Merge branch 'drm-core-next' into drm-linus Merge remote branch 'korg/drm-radeon-next' into drm-linus drm/kms/fb: check for depth changes from userspace for resizing. drm: reduce WARN_ON to a printk. Merge remote branch 'korg/drm-radeon-next' into drm-linus Merge branch 'for-airlied' of /ssd/git/drm-nouveau-next into drm-linus drm/radeon/kms: only evict to GTT if CP is ready drm: fix crtc no modes printf + typo Merge remote branch 'korg/drm-radeon-next' into drm-linus Dave Anderson (1): cgroups: fix 2.6.32 regression causing BUG_ON() in cgroup_diput() Dave Chinner (11): xfs: kill some warnings on i386 builds xfs: Don't flush stale inodes xfs: Ensure we force all busy extents in range to disk lib: Introduce generic list_sort function xfs: reclaim inodes under a write lock xfs: Avoid inodes in reclaim when flushing from inode cache xfs: reclaim all inodes by background tree walks xfs: Remove inode iolock held check during allocation xfs: fix stale inode flush avoidance xfs: fix missing error check in xfs_rtfree_range xfs: xfs_swap_extents needs to handle dynamic fork offsets Dave Jones (1): remove my email address from checkpatch. Dave Liu (1): ucc_geth: Fix the wrong the Rx/Tx FIFO size Dave Young (1): drm: change drm set mode messages as DRM_DEBUG David Daney (3): MIPS: Cleanup forgotten label_module_alloc in tlbex.c MIPS: Octeon: Add sched_clock() to csrc-octeon.c MIPS: Octeon: Use non-overflowing arithmetic in sched_clock David Howells (12): ext4: Don't ask about supporting ext2/3 in ext4 if ext4 is not configured mn10300: wire up missing new syscalls mn10300: use KERN_ERR not KERN_ERROR mn10300: insert PCI root bus resources for the ASB2305 devel motherboard mn10300: make the ASB2305's PCnet32 NIC work by using the PCI bridge's SRAM mn10300: update the ASB2303 defconfig nommu: fix SYSV SHM for NOMMU nommu: struct vm_region's vm_usage count need not be atomic nommu: remove a superfluous check of vm_region::vm_usage nommu: don't need get_unmapped_area() for NOMMU nommu: fix race between ramfs truncation and shared mmap nommu: fix shared mmap after truncate shrinkage problems David John (3): PCI: Check the node argument passed to cpumask_of_node drm: Keep disabled outputs disabled after suspend / resume smp_call_function_any(): pass the node value to cpumask_of_node() David S. Miller (10): Merge branch 'master' of git://git.kernel.org/.../linville/wireless-2.6 Merge branch 'master' of git://git.kernel.org/.../linville/wireless-2.6 cxgb3i: Fix flags test. bbc_envctrl: Clean up properly if kthread_run() fails. Merge branch 'master' of /home/davem/src/GIT/linux-2.6/ sparc64: Fix NMI programming when perf events are active. sparc64: Fix Niagara2 perf event handling. Merge branch 'master' of git://git.kernel.org/.../kaber/nf-2.6 Merge branch 'master' of git://git.kernel.org/.../linville/wireless-2.6 ipv6: skb_dst() can be NULL in ipv6_hop_jumbo(). David Sharp (1): ring-buffer: Wrap a list.next reference with rb_list_head() David VomLehn (1): MIPS: PowerTV: Remove extra r4k_clockevent_init() call David Woodhouse (1): V4L/DVB (13716): [Mantis] Bug: incorrect byte swap Denis Kirjanov (2): vxge: use DMA_BIT_MASK instead of plain values. vxge: use pci_dma_mapping_error to test return value Detlef Riekenberg (1): vgaarbiter: fix a typo in the vgaarbiter Documentation Dexuan Cui (3): PCI: support device-specific reset methods PCI: add Intel USB specific reset method PCI: add Intel 82599 Virtual Function specific reset method Divyesh Shah (1): cfq-iosched: Respect ioprio_class when preempting Dmitri Belimov (1): V4L/DVB (13966): DVB-T regression fix for saa7134 cards Dmitry Torokhov (13): Input: speed up suspend/shutdown for PS/2 mice and keyboards Input: serio - do not mark kseriod freezable anymore Input: ff-memless - another fix for signed to unsigned overflow Input: iforce - fix oops on device disconnect Input: matrix-keypad - handle cases when GPIOs can't be wakeup sources Input: lifebook - add CONFIG_DMI dependency dell-wmi - fix condition to abort driver loading Input: iforce - wait for command completion when closing the device Input: atkbd - fix canceling event_work in disconnect Input: xbox - do not use GFP_KERNEL under spinlock Input: add compat support for sysfs and /proc capabilities output eeepc-laptop: switch to using sparse keymap library sony-laptop - fix using of uninitialized variable Dominik Geyer (1): ath9k: Fix Kconfig depends for ATH9K_DEBUGFS Don Skidmore (1): ixgbe: fix Need to call pci_save_state after pci_restore_state Duane Grigsby (1): [SCSI] qla2xxx: Added to EEH support. Ed Lin (1): [SCSI] stex: fix scan of nonexistent lun Eli Cohen (1): mlx4_core: Fix cleanup in __mlx4_init_one() error path Elliott Sales de Andrade (1): Input: i8042 - add Dritek quirk for Acer Aspire 5610. Emese Revfy (1): drbd: Constify struct file_operations Erez Zadok (5): ecryptfs: initialize private persistent file before dereferencing pointer ecryptfs: don't ignore return value from lock_rename ecryptfs: remove unnecessary d_drop calls in ecryptfs_link ecryptfs: pass matching flags to interpose as defined and used there ecryptfs: fix interpose/interpolate typos in comments Eric Anholt (2): drm/i915: In the debugfs interface, unmap our address instead of the page's. drm/i915: Remove chatty execbuf failure message. Eric Miao (4): [ARM] pxafb: fix building issue of incorrect reference [ARM] pxa/poodle: fix incorrect 'gpio_card_detect' of MMC [ARM] pxa: update pwm_backlight->notify() to include missed 'struct device *' [ARM] pxa: fix compiler warnings of unused variable 'id' in cpu_is_pxa9*() Eric Paris (2): inotify: do not reuse watch descriptors inotify: only warn once for inotify problems Eric Sandeen (2): fs-writeback: Add helper function to start writeback if idle ext4: flush delalloc blocks when space is low Eric W. Biederman (2): sysfs: Add lockdep annotations for the sysfs active reference Input: serio - fix potential deadlock when unbinding drivers Erik Andren (1): V4L/DVB (13880): gspca - m5602-s5k4aa: Add vflip quirk for the Amilo Xi 2428 Erik Andrén (1): V4L/DVB (13882): gspca - stv06xx-vv6410: Ensure register STV_SCAN_RATE is zero Erik-Jan Post (2): viafb: do modesetting after updating variables viafb: fix acceleration for some chips Eugeni Dodonov (1): Staging: asus_oled: fix oops in 2.6.32.2 FUJITA Tomonori (4): block: remove Documentation/block/as-iosched.txt x86/agp: Fix agp_amd64_init() initialization with CONFIG_GART_IOMMU enabled x86/agp: Fix agp_amd64_init and agp_amd64_cleanup powerpc/pseries: Fix dlpar compile warning without CONFIG_PROC_DEVICETREE Fang Wenqi (1): ext4: Update documentation to correct the inode_readahead_blks option name Felipe Balbi (3): Input: twl4030_keypad - switch to using threaded IRQ Input: twl4030-pwrbutton - switch to using threaded IRQ OMAP2xxx clock: clk2xxx.c doesn't compile if CPUFREQ is enabled Felix Fietkau (4): mac80211: fix ibss join with fixed-bssid ath9k: fix missed error codes in the tx status check mac80211: add missing sanity checks for action frames mac80211: fix queue selection for data frames on monitor interfaces Florian Fainelli (3): ipvs: ip_vs_wrr.c: use lib/gcd.c .gitignore: ignore vmlinuz MIPS: BCM63xx: Fix whitespace damaged board_bcm963xx.c Florian Tobias Schandinat (1): viafb: fix LCD hardware cursor regression Florian Westphal (1): netfilter: ebtables: enforce CAP_NET_ADMIN Francisco Jerez (13): drm/nouveau: Add cache_flush/pull fifo engine functions. drm/nouveau: Pre-G80 tiling support. drm/nouveau: Make the MM aware of pre-G80 tiling. drm/i2c/ch7006: Drop build time dependency to nouveau. drm/nouveau: Fix "general protection fault" in the flipd/flips eviction path. drm/nouveau: No need to force evict=true when swapping evicted BOs back in. drm/nouveau: Drop redundant placement initialization. drm/nouveau: Clean up the nv17-nv4x load detection code a bit. drm/nouveau: Implement nv42-nv43 TV load detection. drm/nouveau: Don't skip card take down on nv0x. drm/nouveau: Allocate a per-channel instance of NV_SW. drm/nouveau: Use the software object for fencing. drm/nv04: Context switching fixes. Frederic Weisbecker (22): Merge commit 'v2.6.32' into reiserfs/kill-bkl reiserfs: Fix possible recursive lock reiserfs: Fix reiserfs lock and journal lock inversion dependency reiserfs: Fix reiserfs lock <-> inode mutex dependency inversion reiserfs: Fix remaining in-reclaim-fs <-> reclaim-fs-on locking inversion perf: Pass appropriate frame pointer to dump_trace() reiserfs: Fix reiserfs lock <-> i_xattr_sem dependency inversion reiserfs: Warn on lock relax if taken recursively reiserfs: Fix reiserfs lock <-> i_mutex dependency inversion on xattr reiserfs: Relax reiserfs lock while freeing the journal reiserfs: Relax lock before open xattr dir in reiserfs_xattr_set_handle() reiserfs: Fix unwanted recursive reiserfs lock in reiserfs_unlink() reiserfs: Fix journal mutex <-> inode mutex lock inversion reiserfs: Safely acquire i_mutex from reiserfs_for_each_xattr reiserfs: Safely acquire i_mutex from xattr_rmdir reiserfs: Fix mistake in down_write() conversion reiserfs: Fix recursive lock on lchown reiserfs: Relax the lock before truncating pages reiserfs: Relax lock on xattr removing reiserfs: Don't call reiserfs_get_acl() with the reiserfs lock reiserfs: Relax reiserfs_xattr_set_handle() while acquiring xattr locks perf: Stop stack frame walking off kernel addresses boundaries Gertjan van Wingerde (7): rt2x00: Fix rt2800usb detection in rt2800lib. mac80211: Add define for TX headroom reserved by mac80211 itself. rt2x00: Disable powersaving for rt61pci and rt2800pci. rt2x00: Fix calculation of rt2800 iveiv entry offset. rt2x00: Add USB ID for Linksys WUSB 600N rev 2. rt2x00: Fix LED configuration setting for rt2800. rt2x00: Properly request tx headroom for alignment operations. Giridhar Malavali (1): [SCSI] qla2xxx: Update version number to 8.03.01-k9. Grant Likely (1): powerpc/5200: update defconfigs Greg Kroah-Hartman (4): Revert "sysdev: fix prototype for memory_sysdev_class show/store functions" tty: fix race in tty_fasync Staging: hv: fix smp problems in the hyperv core code USB: add speed values for USB 3.0 and wireless controllers Greg Ungerer (1): m68knommu: fix definitions of __pa() and __va() Guennadi Liakhovetski (7): sh: fix DMA driver's descriptor chaining and cookie assignment ALSA: Fix indentation in pcm_native.c mx3: add support for the mt9v022 camera sensor to pcm037 platform ASoC: fix a memory-leak in wm8903 ASoC: add simplified versions of widget macros ASoC: add helper macros to declare struct soc_enum instances ASoC: add DAI and platform / DMA drivers for SH SIU Gui Jianfeng (1): block: blk_rq_err_sectors cleanup H Hartley Sweeten (5): drivers/block/mg_disk.c: use resource_size() [ARM] pxa: use resource_size() in pwm.c Documentation: fix ioremap return type DocBook: fix ioremap return type ACPI: Remove unnecessary cast. H. Peter Anvin (1): x86, compress: Force i386 instructions for the decompressor Hans Verkuil (1): MAINTAINERS: Andy Walls is the new ivtv maintainer Hans de Goede (1): feature-removal-schedule: Add v4l1 drivers obsoleted by gspca sub drivers Haojian Zhuang (3): [ARM] pxa/zylonite: simplify reduntant gpio settings on mmc slot [ARM] pxa: do not enable L2 after MMU is enabled [ARM] pxa: enable L2 if present in XSC3 Heiko Carstens (19): KVM: get rid of kvm_create_vm() unused label warning on s390 kprobes: Fix distinct type warning x86: copy_from_user() should not return -EFAULT [S390] dasd: add missing compat ptr conversion [S390] cio: add missing compat ptr conversion [S390] vmcp: add missing compat ptr conversion [S390] fs3270: add missing compat ptr conversion [S390] smp: setup smp_processor_id early [S390] smp: remove volatile type quilifier from __cpu_logical_map [S390] tape_block: remove ioctl function [S390] Move __cpu_logical_map to smp.c [S390] bug: implement arch specific __WARN macro [S390] mmap: add missing compat_ptr conversion to both mmap compat syscalls [S390] dasd: add proper compat pointer conversion for symmetrix ioctl [S390] con3215: remove empty ioctl function [S390] unwire sys_recvmmsg again [S390] zcrypt: add sanity check before copy_from_user() [S390] tape_char: add missing compat_ptr conversion driver-core: fix devtmpfs crash on s390 Henrik Rydberg (1): Input: bcm5974 - report ABS_MT events Henrique de Moraes Holschuh (5): thinkpad-acpi: don't take the first ALSA slot by default thinkpad-acpi: don't fail to load the entire module due to ALSA problems thinkpad-acpi: make volume subdriver optional thinkpad-acpi: update volume subdriver documentation thinkpad-acpi: improve Kconfig help text Herbert Xu (1): hwrng: core - Fix double unlock in rng_dev_read Hidetoshi Seto (1): PCI: pcie portdrv: style cleanup Huang Weiyi (4): ext4: remove unused #include <linux/version.h> drbd: remove duplicated #include drbd: remove unused #include <linux/version.h> OMAP2: remove duplicated #include Hugh Dickins (1): mm: move sys_mmap_pgoff from util.c Hui Zhu (1): markup_oops.pl: fix error with x86 Ian Campbell (2): xen: fix hang on suspend. x86: xen: 64-bit kernel RPL should be 0 Ian Molton (1): mfd: tmio_mmc hardware abstraction for CNF area Ingo Molnar (4): Merge branch 'iommu/fixes' of git://git.kernel.org/.../joro/linux-2.6-iommu into x86/urgent ACPI: fix ACPI=n allmodconfig build dma-debug: Fix bug causing build warning Merge branch 'tip/tracing/urgent' of git://git.kernel.org/.../rostedt/linux-2.6-trace into tracing/urgent J. Bruce Fields (1): nfsd: fix "insecure" export option Jack Morgenstein (1): IB/mlx4: Initialize SRQ scatter entries when creating an SRQ Jack Steiner (1): x86, uv: Add function retrieving node controller revision number Jamal Hadi Salim (1): net: restore ip source validation James Bottomley (1): libsrp: fix compile failure James Smart (6): [SCSI] lpfc 8.3.7: Fix FC protocol errors [SCSI] lpfc 8.3.7: Fix NPIV operation errors [SCSI] lpfc 8.3.7: Fix hardware/SLI relates issues [SCSI] lpfc 8.3.7: Fix SCSI protocol related errors. [SCSI] lpfc 8.3.7: Fix discovery failures. [SCSI] lpfc 8.3.7: Update Driver version to 8.3.7 Jamie Iles (1): ARM: 5866/1: arm ptrace: use unsigned types for kernel pt_regs Jan Dumon (6): hso: Add Vendor/Product ID's for new devices hso: Fix for endian issues on big endian machines hso: don't change the state of a closed port hso: Attempt to recover from usb bus errors hso: Fix for 5 sec timeouts with v2.x firmware hso: fixed missing newlines Jan Engelhardt (1): btrfs: fix missing last-entry in readdir(3) Jan Glauber (1): [S390] qdio: convert global statistics to per-device stats Jan Kara (1): quota: Fix dquot_transfer for filesystems different from ext4 Jan Kiszka (2): KVM: x86: Extend KVM_SET_VCPU_EVENTS with selective updates tracing/x86: Derive arch from bits argument in recordmcount.pl Jani Nikula (1): gpiolib: fix poll(2) support reconfigure on sysfs polarity change Janusz Krzysztofik (1): omap: McBSP: Fix possible port lockout Jarek Poplawski (3): net/via-rhine: Fix scheduling while atomic bugs af_packet: Don't use skb after dev_queue_xmit() sky2: Fix oops in sky2_xmit_frame() after TX timeout Jari Vanhala (2): Input: ff-memless - start playing FF effects immediately Input: ff-memless - add notion of direction to for rumble effects Jarkko Lavinen (1): mmc_block: fix probe error cleanup bug Jaroslav Kysela (1): Merge branch 'master' of git://git.kernel.org/.../torvalds/linux-2.6 into fixes Jason Wessel (2): maccess,probe_kernel: Allow arch specific override probe_kernel_(read|write) blackfin,kgdb,probe_kernel: Cleanup probe_kernel_read/write Jaswinder Singh Rajput (1): writeback: add missing kernel-doc notation Jean-Francois Moine (4): V4L/DVB (13816): gspca - main: Set the current frame pointer when first qbuf. V4L/DVB (13622): gspca - ov534: Fix a compilation warning. V4L/DVB (13875): gspca - vc032x: Fix a possible crash with the vc0321 bridge. V4L/DVB (13900): gspca - sunplus: Fix bridge exchanges. Jeff Layton (2): cifs: NULL out tcon, pSesInfo, and srvTcp pointers when chasing DFS referrals sunrpc: on successful gss error pipe write, don't return error Jeff Mahoney (1): virtio: fix section mismatch warnings Jens Axboe (3): Merge branch 'for-jens' of git://git.drbd.org/linux-2.6-drbd into for-linus Merge branch 'for-jens' of git://git.drbd.org/linux-2.6-drbd into for-linus Merge branch 'for-jens' of http://git.drbd.org/linux-2.6-drbd into for-linus Jerome Glisse (4): drm/radeon/kms: Schedule host path read cache flush through the ring V2 drm/radeon/kms: Make sure we release AGP device if we acquired it drm: Avoid calling vblank function is vblank wasn't initialized drm/radeon/kms: Don't try to enable IRQ if we have no handler installed Jesse Barnes (4): drm/i915: only enable hotplug for detected outputs drm/i915: execbuf2 support drm/i915: remove render reclock support drm/i915: disable LVDS downclock by default Jie Zhang (1): NOMMU: Use copy_*_user_page() in access_process_vm() Jiri Slaby (12): PCI: fix section mismatch on update_res() SECURITY: selinux, fix update_rlimit_cpu parameter resource: move kernel function inside __KERNEL__ resource: add helpers for fetching rlimits [IA64] use helpers for rlimits drm/radeon/kms: fix memory leak NET: atlx, fix memory leak reiserfs: Fix unreachable statement [S390] use helpers for rlimits powerpc: Use helpers for rlimits MAINTAINERS: ivtv-devel is moderated Btrfs, fix memory leaks in error paths Jiro SEKIBA (1): nilfs2: trivial coding style fix Joakim Tjernlund (3): zlib: optimize inffast when copying direct from output powerpc/8xx: Always pin kernel instruction TLB powerpc/8xx: Fix user space TLB walk in dcbX fixup Joe Perches (1): scripts/get_maintainer.pl: fix file exclusion X: logic Joerg Roedel (2): x86/amd-iommu: Fix initialization failure panic Merge branches 'dma-debug/fixes' and 'amd-iommu/fixes' into iommu/fixes Johan Hovold (1): usb: serial: fix memory leak in generic driver Johannes Berg (10): iwlwifi: fix EEPROM/OTP reading endian annotations and a bug iwlwifi: fix more eeprom endian bugs mac80211: fix peer HT capabilities mac80211: fix WMM AP settings application wireless: remove remaining qual code cfg80211: fix race between deauth and assoc response cfg80211: fix error path in cfg80211_wext_siwscan mac80211: fix skb buffering issue mac80211: fix endian error cfg80211: fix refcount imbalance when wext is disabled Johannes Thoma (2): drbd: Kconfig fix drbd: Don't go into StandAlone mode when authentification failes because of network error John Fastabend (1): pktgen: ndo_start_xmit can return NET_XMIT_xxx values John Kacur (1): sony_pi: Remove the BKL from open and ioctl John Stultz (1): m68knommu: fix invalid flags on coldfire pit clocksource John W. Linville (3): Revert "b43: Enforce DMA descriptor memory constraints" mac80211: fix-up build breakage in 2.6.33 Revert "mac80211: replace netif_tx_{start,stop,wake}_all_queues" Jonathan Nieder (1): kbuild: really fix bzImage build with non-bash sh Josef Bacik (2): Btrfs: fix regression in orphan cleanup Btrfs: fix possible panic on unmount Julia Lawall (12): drivers/net/wireless: Correct code taking the size of a pointer drivers/block/DAC960.c: use DAC960_V2_Controller drivers/dma: drop unnecesary memset drivers/dma: Correct use after free ext4: Eliminate potential double free on error path drivers/isdn: eliminate duplicated test drivers/net/can: Correct NULL test drivers/net : Correct the size argument to kzalloc drivers/block/drbd: Correct NULL test drivers/block/drbd/drbd_receiver.c: correct NULL test MIPS: Alchemy: Correct code taking the size of a pointer ecryptfs: Eliminate useless code KOSAKI Motohiro (4): proc: partially revert "procfs: provide stack information for threads" futexes: Remove rw parameter from get_futex_key() vmscan: kswapd: don't retry balance_pgdat() if all zones are unreclaimable page allocator: update NR_FREE_PAGES only when necessary Kamalesh Babulal (1): powerpc/hvc: Driver build breaks with !HVC_CONSOLE Kazuhisa Ichikawa (1): mm/page_alloc: fix the range check for backward merging Ken Kawasaki (1): pcnet_cs: add cis of KTI PE520 pcmcia network card Kevin Hilman (1): OMAP3: clock: add clockdomains for UART1 & 2 Kevin Winchester (1): agp: correct missing cleanup on error in agp_add_bridge Kirill Afonshin (1): block: removed unused as_io_context Kristian Høgsberg (3): drm/i915: Move PCI IDs into i915 driver drm/i915: Implement IS_* macros using static tables drm/i915: Track whether cursor needs physical address in intel_device_info Krzysztof Halasa (1): dma-debug: allow DMA_BIDIRECTIONAL mappings to be synced with DMA_FROM_DEVICE and Krzysztof Helt (2): sbawe: fix memory detection part 2 ALSA: ac97: add AC97 STMicroelectronics' codecs Kumar Gala (1): powerpc: 2.6.33 update of defconfigs for embedded 6xx/7xxx, 8xx, 8xxx Kunal Gangakhedkar (1): ALSA: hda - Fix mute led GPIO on HP dv-series notebooks Kuninori Morimoto (1): sh: ms7724: Correct sh-eth EEPROM polling timeout. Kusanagi Kouichi (1): Documentation: Rename Documentation/DMA-mapping.txt Lai Jiangshan (3): tracing/kprobe: Show sign of fields in trace_kprobe format files tracing/syscalls: Fix typo in SYSCALL_DEFINE0 tracing: Fix sign fields in ftrace_define_fields_##call() Lars Ellenberg (1): drbd: check on CONFIG_LBDAF, not LBD Laurent Pinchart (3): V4L/DVB (13826): uvcvideo: Fix controls blacklisting V4L/DVB (13829): uvcvideo: Fix alternate setting selection in isochronous mode V4L/DVB (13831): uvcvideo: Fix oops caused by a race condition in buffer dequeuing Len Brown (23): dell-wmi: sys_init_module: 'dell_wmi'->init suspiciously returned 21, it should follow 0/-E convention ACPI: hp-wmi, msi-wmi: clarify that wmi_install_notify_handler() returns an acpi_status dell-wmi, hp-wmi, msi-wmi: check wmi_get_event_data() return value Merge branch 'wmi' into release Merge branch 'thinkpad' into release Merge branch 'bugzilla-13671' into release Merge branch 'video-error-case' into release dell-wmi: sys_init_module: 'dell_wmi'->init suspiciously returned 21, it should follow 0/-E convention Merge branch 'resume-sci-force-bootparam' into release x86, ACPI: delete acpi_boot_table_init() return value ACPI: enable C2 and Turbo-mode on Nehalem notebooks on A/C ACPI: allow C3 > 1000usec ACPI: delete acpi_processor_power_verify_c2() Merge branch 'acpi-pad' into release Merge branch 'bugzilla-13577-video' into release Merge branch 'bugzilla-14483' into release Merge branch 'bugzilla-14858' into release Merge branch 'bugzilla-15064' into release Merge branch 'ec' into release Merge branch 'eeepc-laptop' into release Merge branch 'osc-bugfix' into release Merge branch 'misc' into release Merge branch 'bugzilla-14954' into release Lennert Buytenhek (2): mwl8k: fix configure_filter() memory leak on error mac80211: fix queue selection for packets injected via monitor interface Li Jie (3): ARM: 5863/1: fix bugs of clock source of NUC900 ARM: 5864/1: Implement arch_reset() in NUC900 ARM: 5865/1: nuc900 ethernet driver needs mii Li Zefan (11): ksym_tracer: Fix to make the tracer work ksym_tracer: Fix to allow writing newline to ksym_trace_filter ksym_tracer: Fix race when incrementing count ksym_tracer: Remove trace_stat ftrace: Fix MATCH_END_ONLY function filter tracing/filters: Fix MATCH_FRONT_ONLY filter matching tracing/filters: Fix MATCH_END_ONLY filter matching lib: Introduce strnstr() tracing/filters: Fix MATCH_MIDDLE_ONLY filter matching tracing/filters: Fix MATCH_FULL filter matching for PTR_STRING tracing/filters: Add comment for match callbacks Linus Torvalds (107): fasync: split 'fasync_helper()' into separate add/remove functions Merge git://git.kernel.org/.../herbert/crypto-2.6 Merge git://git.kernel.org/.../davem/net-2.6 Merge branch 'for-linus' of git://neil.brown.name/md Merge branch 'for-linus' of git://git.kernel.dk/linux-2.6-block Merge branch 'for-linus' of git://git.kernel.org/.../tiwai/sound-2.6 Merge branch 'kvm-updates/2.6.33' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'for-linus' of git://git.kernel.org/.../jbarnes/pci-2.6 Merge branch 'slab/urgent' of git://git.kernel.org/.../penberg/slab-2.6 Merge branch 'for_linus' of git://git.kernel.org/.../tytso/ext4 Merge branch 'fixes' of git://git.kernel.org/.../djbw/async_tx Merge branch 'release' of git://git.kernel.org/.../lenb/linux-acpi-2.6 Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge branch 'tracing-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'perf-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'x86-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'perf-fixes-for-linus-2' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'x86-fixes-for-linus-2' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'for-linus' of git://git.kernel.org/.../ieee1394/linux1394-2.6 Merge git://git.kernel.org/.../sfrench/cifs-2.6 pci: avoid compiler warning in quirks.c twl4030-irq.c: fix compiler warning due to raw-spinlock conversion Merge branch 'reiserfs/kill-bkl' of git://git.kernel.org/.../frederic/random-tracing Merge branch 'for-linus' of git://git.kernel.org/.../ryusuke/nilfs2 Merge branch 'for_linus' of git://git.kernel.org/.../tytso/ext4 Merge branch 'sh/for-2.6.33' of git://git.kernel.org/.../lethal/sh-2.6 Merge branch 'for-linus' of git://git390.marist.edu/pub/scm/linux-2.6 Merge branch 'for-linus/samsung' of git://git.fluff.org/bjdooks/linux Merge branch 'limits_cleanup' of git://decibel.fi.muni.cz/~xslaby/linux Merge branch 'for-2.6.33' of git://git.kernel.org/.../broonie/sound-2.6 Merge branch 'upstream-linus' of git://git.kernel.org/.../jlbec/ocfs2 Merge git://git.kernel.org/.../davem/sparc-2.6 Linux 2.6.33-rc3 Merge master.kernel.org:/home/rmk/linux-2.6-arm Merge branch 'for-linus' of git://git.open-osd.org/linux-open-osd Merge git://git.kernel.org/.../jejb/scsi-rc-fixes-2.6 Merge branch 'davinci-fixes' of git://git.kernel.org/.../khilman/linux-davinci Merge branch 'for-2.6.33' of git://linux-nfs.org/~bfields/linux Merge branch 'drm-intel-next' of git://git.kernel.org/.../anholt/drm-intel Merge branch 'drm-linus' of git://git.kernel.org/.../airlied/drm-2.6 Merge branch 'release' of git://git.kernel.org/.../aegl/linux-2.6 Merge branch 'release' of git://git.kernel.org/.../lenb/linux-acpi-2.6 Merge branch 'bugfixes' of git://git.linux-nfs.org/projects/trondmy/nfs-2.6 Merge branch 'for-linus' of git://git.kernel.org/.../jbarnes/pci-2.6 Merge branch 'x86-fixes-for-linus' of git://git.kernel.org/.../x86/linux-2.6-tip Merge branch 'for-linus/samsung' of git://git.fluff.org/bjdooks/linux Merge branch 'release' of git://git.kernel.org/.../aegl/linux-2.6 Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs Merge branch 'reiserfs/kill-bkl' of git://git.kernel.org/.../frederic/random-tracing Merge branch 'for_linus' of git://git.kernel.org/.../jwessel/linux-2.6-kgdb Merge branch 'for-linus' of git://git.kernel.org/.../roland/infiniband Merge branch 'omap-fixes-for-linus' of git://git.kernel.org/.../tmlind/linux-omap-2.6 Merge branch 'hwmon-for-linus' of git://git.kernel.org/.../jdelvare/staging Merge branch 'drm-linus' of git://git.kernel.org/.../airlied/drm-2.6 Merge branch 'agp-fixes' of git://git.kernel.org/.../airlied/agp-2.6 Merge branch 'for_linus' of git://git.kernel.org/.../jack/linux-fs-2.6 Merge git://git.kernel.org/.../steve/gfs2-2.6-fixes Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs Merge git://git.kernel.org/.../davem/net-2.6 Merge branch 'for-linus' of git://git.kernel.org/.../tiwai/sound-2.6 Merge branch 'upstream' of git://ftp.linux-mips.org/pub/scm/upstream-linus Merge master.kernel.org:/home/rmk/linux-2.6-arm Merge branch 'upstream-linus' of git://git.kernel.org/.../jgarzik/libata-dev Merge branch 'for-linus' of git://gitorious.org/linux-omap-dss2/linux Merge git://git.infradead.org/battery-2.6 Linux 2.6.33-rc4 Merge branch 'fasync-helper' Merge branch 'for-33' of git://repo.or.cz/linux-kbuild Merge branch 'for-linus' of git://git.kernel.org/.../jikos/hid Merge branch 'drm-linus' of git://git.kernel.org/.../airlied/drm-2.6 Merge branch 'for-linus/bugfixes' of git://xenbits.xensource.com/people/ianc/linux-2.6 Merge git://git.kernel.org/.../davem/net-2.6 Merge branch 'for-linus' of git://git390.marist.edu/pub/scm/linux-2.6 Merge branch 'sh/for-2.6.33' of git://git.kernel.org/.../lethal/sh-2.6 Merge branch 'for-linus' of git://git.kernel.org/.../mattst88/alpha-2.6 Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge branch 'for-linus' of git://git.kernel.org/.../bp/bp Merge branch 'for-linus/samsung' of git://git.fluff.org/bjdooks/linux Merge branch 'for-linus' of git://git.kernel.org/.../anholt/drm-intel Merge branch 'tracing-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'perf-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'core-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'x86-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'i2c-for-linus' of git://git.kernel.org/.../jdelvare/staging Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs-2.6 Merge branch 'for_linus' of git://git.kernel.org/.../mchehab/linux-2.6 Merge branch 'mantis' of git://git.kernel.org/.../mchehab/linux-2.6 Merge branch 'for-linus' of git://oss.sgi.com/xfs/xfs Merge branch 'merge' of git://git.kernel.org/.../benh/powerpc Merge branch 'for-linus' of git://git.kernel.org/.../sameo/mfd-2.6 Merge branch 'for-linus' of git://git.monstr.eu/linux-2.6-microblaze Merge branch 'for-linus' of git://git.kernel.org/.../tiwai/sound-2.6 Merge git://git.kernel.org/.../wim/linux-2.6-watchdog Merge master.kernel.org:/home/rmk/linux-2.6-arm Merge branch 'sh/for-2.6.33' of git://git.kernel.org/.../lethal/sh-2.6 Merge git://git.kernel.org/.../mason/btrfs-unstable Merge branch 'for-linus' of git://git.kernel.org/.../ecryptfs/ecryptfs-2.6 Merge branch 'release' of git://git.kernel.org/.../lenb/linux-acpi-2.6 Merge branch 'for-linus' of git://git.kernel.dk/linux-2.6-block Merge git://git.kernel.org/.../gregkh/usb-2.6 Merge git://git.kernel.org/.../gregkh/driver-core-2.6 Merge git://git.kernel.org/.../gregkh/staging-2.6 Merge git://git.kernel.org/.../gregkh/tty-2.6 Merge branch 'upstream-linus' of git://git.kernel.org/.../jgarzik/libata-dev Merge branch 'sched-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Merge branch 'perf-fixes-for-linus' of git://git.kernel.org/.../tip/linux-2.6-tip Linux 2.6.33-rc5 Linus Walleij (1): ARM: 5867/1: Update U300 defconfig Lothar Wassmann (2): USB: isp1362: better 64bit printf warning fixes USB: isp1362: fix build failure on ARM systems via irq_flags cleanup Luca Barbieri (2): drm/nouveau: Fix null deref in nouveau_fence_emit due to deleted fence hw-breakpoints, perf: Fix broken mmiotrace due to dr6 by reference change Luca Tettamanti (3): drm/radeon/kms: rs600: use correct mask for SW interrupt hwmon: (asus_atk0110) Refactor interface probe code hwmon: (asus_atk0110) Add debugfs interface Luis R. Rodriguez (6): ath9k: wake hardware for interface IBSS/AP/Mesh removal ath9k: wake hardware during AMPDU TX actions mac80211: fix race with suspend and dynamic_ps_disable_work mac80211: fix propagation of failed hardware reconfigurations ath5k: Fix eeprom checksum check for custom sized eeproms cfg80211: fix syntax error on user regulatory hints Maarten Maathuis (2): drm/nouveau: better alignment of bo sizes and use roundup instead of ALIGN drm/nv50: make the blocksize depend on vram size Magnus Horlin (1): V4L/DVB (13748): [Mantis/VP-2040] Add support for VP-2040 (TDA10023 frontend based) Manu Abraham (101): V4L/DVB (13699): [Mantis, MB86A16] Initial checkin: Mantis, MB86A16 V4L/DVB (13700): [MB86A16] Need a bit of settling time V4L/DVB (13701): [MB86A16] Reduce Carrier Recovery range to 3Mhz V4L/DVB (13702): [MB86A16] need to wait a bit more than the computed time for a Factor of safety V4L/DVB (13703): [MB86A16] Fix wrong message printed out V4L/DVB (13704): [MB86A16] FIX: Don't loop again, if we have SYNC V4L/DVB (13705): [Mantis] FIX: Do not return IRQ_HANDLED in the unlikely case V4L/DVB (13706): [MB86A16] Overhaul V4L/DVB (13707): [Mantis] Whitespace cleanup V4L/DVB (13708): [Mantis] Remove some dead code V4L/DVB (13709): [Mantis/VP-1034] Switch 13/18v for the VP-1034 properly V4L/DVB (13710): [Mantis] FIX: Use swfilter (188/204) accordingly V4L/DVB (13712): [Mantis] Add locking for concurrent access V4L/DVB (13714): [MB86A16] FIX/Code simplification: use hwconfig->ts_size instead of ts_size V4L/DVB (13715): [Mantis] Kernel I2C changes: use PCI parent device V4L/DVB (13717): [MB86A16] Statistics Updates V4L/DVB (13719): [Mantis/VP-2033] Initial test switch to the tda10021, from the cu1216 V4L/DVB (13720): [Mantis/Terratec Cinergy C] Add support for the Terratec Cinergy C PCI V4L/DVB (13721): [Mantis] Bug! Before bailing out, Unlock V4L/DVB (13722): [Mantis] Revert 13560 V4L/DVB (13723): [Mantis/VP-2040, Terratec Cinergy C] Add support for the Cinergy C, VP-2040 clone V4L/DVB (13724): [Mantis/VP-1041] Initial support for Mantis VP-1041 V4L/DVB (13725): [Mantis/VP-1041] Revert to old register initialization parameters, for now. V4L/DVB (13726): [Mantis/Skystar HD2] Add support for the Technisat Skystar HD2 V4L/DVB (13727): [Mantis/VP-1041] Bugfix: Sigh! Don't look for the STOP bit V4L/DVB (13728): [Mantis] Add in some Host Interface definitions V4L/DVB (13729): [Mantis] Add in a license header V4L/DVB (13730): [Mantis] Add in some UART definitions V4L/DVB (13731): [Mantis] Add in a license header V4L/DVB (13732): [Mantis] Add in some Link Layer definitions V4L/DVB (13733): [Mantis] Start with the PCMCIA interface V4L/DVB (13734): [Mantis] Initial go at an Event Manager V4L/DVB (13735): [Mantis] Implement the Event Manager tasklet V4L/DVB (13736): [Mantis] Implement CAM Plug IN and Unplug events V4L/DVB (13737): [Mantis] Register the CA device, dummy functions for now V4L/DVB (13738): [Mantis] Enable IRQ0 events V4L/DVB (13739): [Mantis] Event Manager: Handle Masked events only V4L/DVB (13740): [Mantis] Schedule the work instead of handling the task directly V4L/DVB (13741): [Mantis] Implement HIF Mem Read/Write operations V4L/DVB (13742): [Mantis] Implement PCMCIA I/O Rd/Wr operations V4L/DVB (13743): [Mantis CA] Use DVB_CA Tuple parser V4L/DVB (13744): [Mantis CA] Use Module status to signal Slot events V4L/DVB (13745): [Mantis CA] Add some debug statements V4L/DVB (13746): [Mantis CA] Bug: Remove duplicated symbol V4L/DVB (13749): [Mantis CA] CA_SLAVE: Do not change Slave Configuration setup V4L/DVB (13750): [Mantis] GPIO_CONTROL: Cache a given GPIO Bit Setup for a given event V4L/DVB (13751): [Mantis] GPIO_CONTROL: Do not toggle GPIO CW's on HIF operations V4L/DVB (13752): [Mantis CA] CAM_CONTROL: All CAM control operations now handled by the worker thread V4L/DVB (13753): [Mantis CA] SLOT_CONTROL: Implement Slot RESET V4L/DVB (13754): [Mantis] CAM_CONTROL: Implement TS control V4L/DVB (13755): [Mantis CA] CAM_CONTROL: Use appropriate flags V4L/DVB (13756): [Mantis CA] CAM_CONTROL: Use CAMCHANGE_IRQ events V4L/DVB (13757): [Mantis CA] CAM_CONTROL: Use FRDA_IRQ Events V4L/DVB (13758): [Mantis CA] CAM_CONTROL: Use CAMREADY_IRQ event V4L/DVB (13759): [Mantis] HIF I/O: Use the LSB octet only V4L/DVB (13761): [Mantis] HIF I/O: Temporary workaround, use SBUF_OPDONE flag instead V4L/DVB (13762): [Mantis CA] CA_MODULE: Look for the module status on driver unload as well V4L/DVB (13763): [Mantis] HIF I/O: trim delays a bit appropriately V4L/DVB (13764): [Mantis CA] SLOT: Add some debug status V4L/DVB (13765): [Mantis] HIF I/O: Add some debug statements V4L/DVB (13767): [Mantis/VP-1041] Bug: Add in missing Master clock settings V4L/DVB (13768): [Mantis] Enable WRACK V4L/DVB (13769): [Mantis] Smart Buffer Burst Read Ready cannot flag FR/DA Irq V4L/DVB (13770): [Mantis] Bug Do not trigger FR/DA IRQ from SBUF OPDONE V4L/DVB (13771): [Mantis] Reset Flags at the earliest possible V4L/DVB (13772): [Mantis] Do not enable Common Memory Access V4L/DVB (13773): [Mantis] Enable all interrupts V4L/DVB (13775): [Mantis] Remove unnecessary job queues V4L/DVB (13776): [Mantis] Use a simple timeout instead, interruptible V4L/DVB (13777): [Mantis] Use a Write wait queue for Write events V4L/DVB (13778): [Mantis] Wr ACK is already handled in the fast path, V4L/DVB (13779): [Mantis] Missing wakeup for write queue V4L/DVB (13781): [Mantis CA] Bug: Fix wrong usage of HIFRDWRN V4L/DVB (13782): [Mantis] Temporarily disable FRDA irq V4L/DVB (13784): [Mantis] Use PCI API instead of hardcoded length V4L/DVB (13785): [Mantis] Do not disable IRQ's while being invoked V4L/DVB (13786): [Mantis] Bug: HIF bits already shifted .. V4L/DVB (13787): [Mantis] Fix build V4L/DVB (13788): [Mantis CA] use a lock for the relevant CI Read/Write operations V4L/DVB (13789): [Mantis CA] Initialize the mutex V4L/DVB (13790): [Mantis] Relocate queue initialization V4L/DVB (13794): [Mantis/VP-3028] Initial go at Serial interface implementation, add support for VP-3028 V4L/DVB (13795): [Mantis/Hopper] Code overhaul, add Hopper devices into the PCI ID list V4L/DVB (13796): [Mantis] Add missing file in previous commit V4L/DVB (13797): [Mantis/Hopper/TDA665x] Large overhaul, V4L/DVB (13798): [Mantis] Enable power for all cards, use byte mode only on relevant devices V4L/DVB (13799): [Mantis] Unregister frontend V4L/DVB (13800): [Mantis] I2C optimization. Required delay is much lesser than 1mS. V4L/DVB (13801): [MB86A16] Use the search callback V4L/DVB (13802): [Mantis/Hopper] Fix all build related warnings V4L/DVB (13803): Remove unused dependency on CU1216 V4L/DVB (13804): Remove unused I2C Adapter ID V4L/DVB (13805): Fix: Unregister the frontend before detaching V4L/DVB (13806): Register and Initialize Remote control V4L/DVB (13807): Fix: Free device in the device registration failure case V4L/DVB (13809): Fix Checkpatch violations V4L/DVB (13810): [MB86A16] Use DVB_* macros V4L/DVB (13811): [MB86A16] Update Copyright header V4L/DVB (13812): [Mantis/Hopper] Update Copyright header V4L/DVB (13808): [Mantis/Hopper] Build update for Mantis/Hopper based cards V4L/DVB (13851): Fix Input dependency for Mantis Marc Zyngier (3): [ARM] pxa/zeus: make internal zeus_get_pcb_info static [ARM] pxa/zeus: relax memory timings on Zeus ethernet ports [ARM] pxa/zeus: provide power-source information when APM is enabled Marcelo Tosatti (2): KVM: MMU: remove prefault from invlpg handler KVM: LAPIC: make sure IRR bitmap is scanned after vm load Marcin Kościelnicki (1): drm/nv04: Fix set_operation software method. Marcin Slusarz (2): drm/nv50: fix fillrect color drm/nouveau: create function for "dealing" with gpu lockup Marek Vasut (2): [ARM] pxa/littleton: add UART3 GPIO config [ARM] pxa/littleton: select CPU_PXA300 and CPU_PXA310 Mark Brown (6): cs89x0: Always report failure to request interrupt ARM: SMDK6410: Specify no GPIO for B_PWR_5V regulator revert "drivers/video/s3c-fb.c: fix clock setting for Samsung SoC Framebuffer" mfd: Correct WM835x ISINK ramp time defines mfd: WM835x GPIO direction register is not locked Merge branch 'for-2.6.33' into for-2.6.34 Mark Salter (3): mn10300: signal stack fix mn10300: objcopy flags fix mn10300: add cc clobbers to asm statements Marko Ristola (2): V4L/DVB (13711): [Mantis] FIX: Do nor toggle GPIF status V4L/DVB (13718): [Mantis] Use gpio_set_bits to turn OFF the bits as well Marko Viitamaki (1): V4L/DVB (13783): [Mantis/Technisat Cablestar HD2] Add support for the Technisat Cablestar HD2 Markus Pietrek (1): serial: sh-sci: using correct fifo size for SCIF and SCIFA ports. Martin K. Petersen (6): block: Fix topology stacking for data and discard alignment block: Fix incorrect alignment offset reporting and update documentation block: Correct handling of bottom device misaligment block: Fix discard alignment calculation and printing block: bdev_stack_limits wrapper DM: Fix device mapper topology stacking Martin Schwidefsky (5): [S390] Update default configuration. [S390] fix loading of PER control registers for utrace. [S390] clear TIF_SINGLE_STEP for new process. [S390] duplicate SIGTRAP on signal delivery. [S390] remove superfluous TIF_USEDFPU bit Masami Hiramatsu (2): x86: Fix objdump version check in chkobjdump.awk for different formats. kmod: fix resource leak in call_usermodehelper_pipe() Masanari Iida (1): ALSA: Fix a typo in Procfile.txt Matt Carlson (5): tg3: Fix std prod ring nicaddr for 5787 and 57765 tg3: Fix std rx prod ring handling tg3: Add reliable serdes detection for 5717 A0 tg3: Disable 5717 serdes and B0 support tg3: Update copyright and driver version Matt Turner (1): alpha: add myself as a maintainer, and drop mention of 2.4 Matthew Garrett (2): drm/i915: Don't check for lid presence when detecting LVDS serial: 8250_pnp: use wildcard for serial Wacom tablets Matthew Slattery (3): sfc: QT2025C: Work around PHY bug sfc: QT2025C: Switch into self-configure mode when not in loopback sfc: QT2025C: Work around PHY firmware initialisation bug Mauro Carvalho Chehab (5): V4L/DVB (13834): dib8000: fix compilation if !DVB_DIB8000 V4L/DVB (13858): ir-keytable: use the right header V4L/DVB(13808a): mantis: convert it to the new ir-core register/unregister functions V4L/DVB (13808b): mantis: replace DMA_nnBIT_MASK to DMA_BIT_MASK(32) V4L/DVB(13824a): mantis: Fix __devexit bad annotations Michael Chan (1): bnx2x: Initialize cnic status block during chip reset Michael Ellerman (1): powerpc/iseries: Initialise on-stack completion Michael Hennerich (1): gpio: adp5588-gpio: new driver for ADP5588 GPIO expanders Michael Hernandez (1): [SCSI] qla2xxx: Get the link data rate explicitly during device resync. Michael Krufky (2): V4L/DVB (13887): tda8290: add autodetection support for TDA8295c2 V4L/DVB (13934): tda8290: Fix FM radio easy programming standard selection for TDA8295 Michal Marek (1): Makefile: do not override LC_CTYPE Michal Simek (5): microblaze: Wire up recvmmsg syscall microblaze: Enable accept4 syscall microblaze: Add PT_ macros for special purpose regs microblaze: Add missing double apostrophe in Kconfig microblaze: pci_controller->arch_data really is a struct device_node * Michel Dänzer (1): drm/radeon/kms: Fix crash getting TV info with no BIOS. Miguel Aguilar (2): Input: davinci_keyscan - add device_enable method to platform data DaVinci: DM365: Add the device_enable for the DaVinci Keyscan Mikael Pettersson (1): sata_promise: don't classify overruns as HSM errors Mike Frysinger (3): FDPIC: Respect PT_GNU_STACK exec protection markings when creating NOMMU stack NOMMU: Avoiding duplicate icache flushes of shared maps modpost: fix segfault in sym_is() with prefixed arches Mike Galbraith (1): sched: Fix vmark regression on big machines Mike Travis (3): x86: SGI UV: Fix writes to led registers on remote uv hubs x86_64 SGI UV: Fix writes to led registers on remote uv hubs. x86: SGI UV: Fix mapping of MMIO registers Minchan Kim (1): smaps: fix wrong rss count Ming Lei (1): ath9k: fix ito64 Márton Németh (5): hwmon: Make PCI device ids constant powerpc/macintosh: Make Open Firmware device id constant i2c-ali1563: Remove sparse warnings acpi: make ACPI device id constant V4L/DVB (13941): rj54n1cb0c: remove compiler warning NISHIMOTO Hiroki (1): sh: mach-ecovec24: Add motion sensor driver support. Nathan Fontenot (2): powerpc: Move /proc/ppc64 to /proc/powerpc update powerpc: Move cpu hotplug driver lock from pseries to powerpc Neil Turton (1): sfc: Fix DMA mapping cleanup in case of an error in TSO NeilBrown (4): md: Fix unfortunate interaction with evms md: fix small irregularity with start_ro module parameter md: remove unnecessary code from do_md_run md: allow a resync that is waiting for other resync to complete, to be aborted. Nemeth Marton (1): V4L/DVB (13820): lgdt3305: make one-bit bitfields unsigned Nicolas Ferre (1): dma: at_hdmac: correct incompatible type for argument 1 of 'spin_lock_bh' Niklas Edmundsson (2): V4L/DVB (13791): [TDA10021] Do not claim TDA10023 V4L/DVB (13792): [Mantis/VP-2033] Do not claim TDA10023 OGAWA Hirofumi (5): block: Honor the gfp_mask for alloc_page() in blkdev_issue_discard() nfs: fix oops in nfs_rename() rtc_cmos: convert shutdown to new pnp_driver->shutdown vfs: Fix vmtruncate() regression ecryptfs: Fix refcnt leak on ecryptfs_follow_link() error path Octavian Purdila (3): ip: fix mc_loop checks for tunnels with multicast outer addresses tcp: update the netstamp_needed counter when cloning sockets net: fix build erros with CONFIG_BUG=n, CONFIG_GENERIC_BUG=n Oliver Neukum (1): USB: Don't use GFP_KERNEL while we cannot reset a storage device Or Gerlitz (1): IB/mlx4: Fix queue overflow check in post_recv PJ Waskiewicz (1): ixgbe: Fix compiler warning about variable being used uninitialized Patrick McHardy (1): netfilter: nf_ct_ftp: fix out of bounds read in update_nl_seq() Paul Mundt (6): sh: Only provide a PCLK definition for legacy CPG CPUs. sh: Disable PMB for SH4AL-DSP CPUs. sh: Don't default enable PMB support. sh: update defconfigs. sh: unwire sys_recvmmsg. sh64: wire up sys_accept4. Paul Rolland (2): wmi: check find_guid() return value to prevent oops wmi: check find_guid() return value to prevent oops Paul Walmsley (7): OMAP2xxx IO mapping: mark DSP mappings as being 2420-only OMAP2420 IO mapping: move IVA mapping virtual address out of vmalloc space OMAP3 clock: McBSP 2, 3, 4 functional clock parent is PER_96M_FCLK, not CORE_96M_FCLK OMAP clock: remove incorrect EXPORT_SYMBOL()s OMAP2xxx OPP: clean up comments in OPP data OMAP clock/CPUFreq: add clk_exit_cpufreq_table() OMAP2 clock: dynamically allocate CPUFreq frequency table Pavel Machek (1): serial: serial_cs: oxsemi quirk breaks resume Pavel Roskin (1): rt2x00: use correct headroom for transmission Pekka Enberg (5): x86: Use KERN_DEFAULT log-level in __show_regs() x86, kmemcheck: Use KERN_WARNING for error reporting SLAB: Fix lockdep annotation breakage perf kmem: Increase "Hit" column length perf kmem: Print usage help for unknown commands Peter Huewe (2): ALSA: sound/arm: Fix build failure caused by missing struct aaci definition video/omap: add __init/__exit macros to drivers/video/omap/lcd_htcherald.c Peter Hüwe (2): ARM: 5870/1: arch/arm: Fix build failure for defconfigs without CONFIG_ISA_DMA_API set ARM: 5871/1: arch/arm: Fix build failure for lpd7a404_defconfig caused by missing includes Peter Tyser (1): powerpc: Use scripts/mkuboot.sh instead of 'mkimage' Peter Zijlstra (4): perf: Fix NULL deref in inheritance code perf: Fix perf_event_do_pending() fallback callsite perf: Honour event state for aux stream data perf: Change the is_software_event() definition Petr Štetiar (1): HID: blacklist ET&T TC5UH touchscreen controller Philipp Reisner (5): drbd: Use drbd_crypto_is_hash() instead of an open coded check drbd: Fix for a race between IO and a detach operation [Bugz 262] drbd: Silenced an assert that could triggered after changing write ordering method drbd: Allow online resizing of DRBD devices while peer not reachable (needs to be explicitly forced) drbd: The kernel code is now equivalent to out of tree release 8.3.7 Ping (1): serial/8250_pnp: add a new Fujitsu Wacom Tablet PC device Przemo Firszt (2): HID: fix pad button definition in hid-wacom HID: wacom: Add BTN_TOOL_FINGER for pad button reporting Rabin Vincent (2): ARM: 5868/1: ARM: fix "BUG: using smp_processor_id() in preemptible code" ARM: 5883/1: Revert "disable NX support for OABI-supporting kernels" Rafael J. Wysocki (5): PCI/PM: Propagate wake-up enable for PCIe devices too PCI: Fix build if quirks are not enabled PCI/PM: Use per-device D3 delays DRM / i915: Fix resume regression on MSI Wind U100 w/o KMS drm/i915: Fix resume regression on MSI Wind U100 w/o KMS Rakesh Ranjan (2): [SCSI] cxgb3i: Fix a login over vlan issue cxgb3i: Fix a login over vlan issue Rakib Mullick (2): Input: wistron - fix test for CONFIG_PM ACPI: Fix unused variable warning in sbs.c Ralf Baechle (1): NET: XFRM: Fix spelling of neighbour. Ramax Lo (1): ARM: S3C: NAND: Check the existence of nr_map before copying Randy Dunlap (11): Documentation: Update mmiotrace.txt Documentation: Update tracepoint-analysis.txt Documentation: Update ftrace-design.txt tracing: Kconfig spelling fixes and cleanups sunrpc: fix build-time warning kgdb: Fix kernel-doc format error in kgdb.h docs: large update to ioctl-number.txt power: fix kernel-doc notation Documentation: update ring-buffer-design.txt documentation: update kernel-doc-nano-HOWTO information tty.h: make tty_port_get() static inline Reinette Chatre (4): iwlwifi: power up all devices for EEPROM read iwl3945: disable power save iwlwifi: initialize spinlock before use iwlwifi: fix 40MHz operation setting on cards that do not allow it René Bolldorf (1): Input: psmouse - fix compile warning in hgpk module Richard Kennedy (1): ext4: return correct wbc.nr_to_write in ext4_da_writepages Robert P. J. Day (2): [S390] Have param.h simply include <asm-generic/param.h>. IB/addr: Correct CONFIG_IPv6 to CONFIG_IPV6 Robert Schedel (1): HID: Support 171 byte variant of Samsung USB IR receiver Roel Kluin (14): drbd: fix test of unsigned in _drbd_fault_random() drbd: Fix test of unsigned in _drbd_fault_random() iwmc3200wifi: Fix test of unsigned in iwm_ntf_stop_resume_tx() ALSA: test off by one in setsamplerate() wl1251: timeout one too soon in wl1251_boot_run_firmware() Input: gf2k - fix &&/|| confusion in gf2k_connect() drm/kms: Fix &&/|| confusion in drm_fb_helper_connector_parse_command_line() omap: &&/|| confusion in iommu_put() omap3: add missing parentheses omap3: add missing parentheses amd64_edac: Ensure index stays within bounds in amd64_get_scrub_rate i2c: Test off by one in {piix4,vt596}_transaction() V4L/DVB (13868): gspca - sn9c20x: Fix test of unsigned. serial: imx: bit &/| confusion Roger Blofeld (1): hwmon: (adt7462) Fix pin 28 monitoring Roland Dreier (2): Merge branches 'misc' and 'mlx4' into for-next kernel.h: add BUILD_BUG_ON_NOT_POWER_OF_2() Rolf Eike Beer (1): kfifo: Fix typo in comment Russ Anderson (1): x86, uv: Ensure hub revision set for all ACPI modes. Russell King (7): Merge branch 'fix' of git://git.kernel.org/.../ycmiao/pxa-linux-2.6 Merge branch 'for-rmk' of git://git.pengutronix.de/git/imx/linux-2.6 ARM: add missing recvmmsg syscall number Merge branch 'fix' of git://git.kernel.org/.../ycmiao/pxa-linux-2.6 ARM: Fix wrong dmb ARM: Ensure ARMv6/7 mm files are built using appropriate assembler options ARM: fix badly placed mach/plat entries in Kconfig & Makefile Rusty Russell (3): lguest: fix bug in setting guest GDT entry Revert "x86: Side-step lguest problem by only building cmpxchg8b_emu for pre-Pentium" Revert "x86: Side-step lguest problem by only building cmpxchg8b_emu for pre-Pentium" Ryan May (1): USB: fix usbstorage for 2770:915d delivers no FAT Ryusuke Konishi (1): nilfs2: update mailing list address Saeed Bishara (1): mv643xx_eth: don't include cache padding in rx desc buffer size Samuel Ortiz (1): libertas: Remove carrier signaling from the scan code Sandeep Gopalpet (1): gianfar: Fix gianfar select_queue bogosity Santosh Shilimkar (1): ARM: 5872/1: ARM: include needed linux/cpu.h in asm/cpu.h Sarah Sharp (1): USB: Fix duplicate sysfs problem after device reset. Sarveshwar Bandi (3): be2net: Bug fix to avoid soft lockup in loopback test. be2net: Bug fix to config NIC appropriately before loopback test be2net: Bug fix to return correct values in ethtool get_settings. Sascha Hauer (1): lib/rational.c needs module.h Sekhar Nori (3): davinci: cp_intc: provide set_wake function davinci: da8xx/omap-l1: mark RTC as a wakeup source davinci: enable ARCH_HAS_HOLES_MEMORYMODEL for DaVinci Serge E. Hallyn (1): generic_permission: MAY_OPEN is not write access Seth Heasley (1): [WATCHDOG] iTCO_wdt: Add Intel Cougar Point and PCH DeviceIDs Shannon Nelson (1): ixgbe: update copyright dates Shaohua Li (1): cfq-iosched: don't regard requests with long distance as close Shaun Ruffell (1): dma-debug: Do not add notifier when dma debugging is disabled. Sheng Yang (1): KVM: Fix possible circular locking in kvm_vm_ioctl_assign_device() Sigmund Augdal (6): V4L/DVB (13713): [MB86A16] Fix: Initialize SNR/STATUS V4L/DVB (13747): [Mantis] Bug Fix!: Use Register Address rather than register field V4L/DVB (13760): [Mantis CA] CA_MODULE: Look for module status on driver load V4L/DVB (13766): [Mantis] Bug: Fix wrong exit condition V4L/DVB (13774): [Mantis] Remove redundant wait for Burst Reads, wakeup the HIF event V4L/DVB (13780): [Mantis] HIF I/O: Enable Interrupts for Read Simon Kagstrom (1): ARM: 5874/1: serial21285: fix disable_irq-from-interrupt-handler deadlock Sonic Zhang (1): blackfin,kgdb: Do not put PC in gdb_regs into retx. Sriram (1): TI DaVinci EMAC: Handle emac module clock correctly. Stanislav Brabec (1): serial-core: resume serial hardware with no_console_suspend Stefan Assmann (2): PCI: change PCI nomenclature in drivers/pci/ (comment changes) PCI: change PCI nomenclature in drivers/pci/ (non-comment changes) Stefan Richter (4): firewire: cdev: fix another memory leak in an error path firewire: ohci: always use packet-per-buffer mode for isochronous reception firewire, ieee1394: update MAINTAINERS entries firewire, ieee1394: update Kconfig help Stefan Roese (1): powerpc: Fix decrementer setup on 1GHz boards Stefani Seibold (2): kfifo: fix kfifo_out_locked race bug USB: serial: fix USB serial fix kfifo_len locking Stephane Eranian (1): perf: x86: Add support for the ANY bit Stephen Hemminger (2): genhd: overlapping variable definition drivers/cpuidle/governors/menu.c: fix undefined reference to `__udivdi3' Steve French (1): [CIFS] Enable mmap on forcedirectio mounts Steve Hodgson (1): sfc: Move PHY software state initialisation from init() into probe() Steven Rostedt (2): tracing: Fix setting tracer specific options ring-buffer: Add rb_list_head() wrapper around new reader page next field Steven Whitehouse (4): GFS2: Ensure uptodate inode size when using O_APPEND GFS2: Fix locking bug in rename GFS2: Fix gfs2_xattr_acl_chmod() GFS2: Use MAX_LFS_FILESIZE for meta inode size Sucheta Chakraborty (2): netxen: fix ethtool register dump netxen: fix ethtool link test Sujith (4): ath9k: Fix bug in assigning sequence number ath9k: Fix TX queue draining ath9k: Stop ANI when doing a reset ath9k: fix suspend by waking device prior to stop Surbhi Palande (1): ext4: replace BUG() with return -EIO in ext4_ext_get_blocks Suresh Siddha (3): x86, irq: Check move_in_progress before freeing the vector mapping x86, apic: use physical mode for IBM summit platforms x86, apic: use logical flat for systems with <= 8 logical cpus Tai-hwa Liang (2): Input: pmouse - move Sentelic probe down the list Input: sentelic - fix left/right horizontal scroll mapping Takashi Iwai (15): ALSA: hda - Disable tigger at pin-sensing on AD codecs ALSA: hda - use snd_hda_jack_detect() again in patch_sigmatel.c ALSA: hda - Don't cache beep controls ALSA: hda - Fix Oops at reloading beep devices Merge branch 'fix/hda' into for-linus ALSA: usb-audio - Avoid Oops after disconnect Merge remote branch 'alsa/fixes' into fix/misc ALSA: hda - Fix ALC861-VD capture source mixer Merge branch 'fix/hda' into for-linus Merge branch 'fix/asoc' into for-linus ALSA: hda - Fix Toshiba NB20x quirk entry ALSA: hda - Fix missing capture mixer for ALC861/660 codecs ALSA: hda - Fix capture on Sony VAIO with single input ALSA: Remove warning message for invalid OSS minor ranges Merge branch 'fix/hda' into for-linus Tamas Vincze (1): edac: i5000_edac critical fix panic out of bounds Tao Ma (1): ocfs2: Handle O_DIRECT when writing to a refcounted cluster. Tejun Heo (3): ata_piix: enable 32bit PIO on SATA piix libata: retry link resume if necessary libata: retry FS IOs even if it has failed with AC_ERR_INVALID Thadeu Lima de Souza Cascardo (2): ALSA: use subsys_initcall for sound core instead of module_init i2c: Do not use device name after device_unregister Theodore Ts'o (5): ext4: add module aliases for ext2 and ext3 ext4, jbd2: Add barriers for file systems with exernal journals ext4: Patch up how we claim metadata blocks for quota purposes ext4: Fix accounting of reserved metadata blocks ext4: Calculate metadata requirements more accurately Thiago Farina (1): fs/bio.c: fix shadows sparse warning Thomas Champagne (1): pmu_battery: Fix battery full reporting Tim Blechmann (1): perf: Rename perf_event_hw_event in design document Tobias Klauser (5): nilfs2: Storage class should be before const qualifier ath9k: Storage class should be before const qualifier iwlwifi: Storage class should be before const qualifier drm/i915: Storage class should be before const qualifier i2c-core: Storage class should be before const qualifier Tomaz Mertelj (1): hwmon: driver for Texas Instruments amc6821 chip Tomi Valkeinen (12): OMAP: DSS2: DSI: fix VC channels in send_short and send_null OMAP: DSS2: DSI: print debug DCS cmd in hex OMAP: DSS2: Collect interrupt statistics OMAP: DSS2: Fix crash when panel doesn't define enable_te() OMAP: DSS2: RFBI: convert to new kfifo API OMAP: OMAPFB: fix clk_get for RFBI OMAP: OMAPFB: add dummy release function for omapdss MAINTAINERS: change omapfb maintainer MAINTAINERS: Combine DSS2 and OMAPFB2 into one entry OMAP: DSS2: OMAPFB: fix omapfb_free_fbmem() OMAP: DSS2: Make check-delay-loops consistent OMAP: DSS2: OMAPFB: fix crash when panel driver was not loaded Tony Lindgren (9): omap: Remove uninitialized warning for gpio.c omap1: Fix compile for omap1_bl.c omap3: Fix booting if package is uninitialized Merge branch 'for_2.6.33rc_c' of git://git.pwsan.com/linux-2.6 into omap-fixes-for-linus ARM: 5882/1: ARM: Fix uncompress code compile for different defines of flush(void) ARM: 5884/1: arm: Fix DCC console for v7 ARM: 5885/1: arm: Flush TLB entries in setup_mm_for_reboot() ARM: 5886/1: arm: Fix cpu_proc_fin() for proc-v7.S and make kexec work ARM: 5888/1: arm: Update comments in cacheflush.h and remove unnecessary V6 and V7 comments Tony Luck (5): KVM: ia64: fix build breakage due to host spinlock change [IA64] sanity in #include files. Move fnptr to types.h [IA64] __per_cpu_idtrs[] is a memory hog [IA64] move fnptr definition inside #ifdef __KERNEL__ ACPI: Fix section mismatch error for acpi_early_processor_set_pdc() Trond Myklebust (2): SUNRPC: Fix up an error return value in gss_import_sec_context_kerberos() SUNRPC: Fix the return value in gss_import_sec_context() Tuukka Toivonen (1): OMAP3 clock: Add capability to change rate of dpll4_m5_ck Tyler Hicks (3): eCryptfs: Use notify_change for truncating lower inodes eCryptfs: Add getattr function eCryptfs: Remove mmap from directory operations Uri Yosef (1): ARM: MINI2440: Fix crash on boot due to improper __initdata qualifier Ursula Braun (1): claw: use "claw" as root device name Uwe Kleine-König (5): [ARM] pxa/ttc_dkb: remove duplicate macro definition imx/mx3: depend on USB_ULPI for otg_ulpi_create vsnprintf: fix reference for compressed ipv6 addresses V4L/DVB mx1_camera: don't check platform_get_irq's return value against zero V4L/DVB sh_mobile_ceu: don't check platform_get_irq's return value against zero Vaibhav Hiremath (3): Davinci VPFE Capture: Take i2c adapter id through platform data omap3: EVM: Choose OMAP_PACKAGE_CBB OMAP: DSS2: Fix compile warning Valentin Longchamp (1): mx31moboard: fix usbh device names Vegard Nossum (1): kmemcheck: make bitfield annotations truly no-ops when disabled Ville Syrjälä (1): OMAP: DSS2: Reject scaling settings when they cannot be supported Vimal Singh (1): omap2/3: ZOOM: Correcting key mapping for few keys Vitaliy Gusev (1): tun: use tun_sk instead container_of Vivek Goyal (3): cfq-iosched: Remove the check for same cfq group from allow_merge cfq-iosched: Get rid of nr_groups cfq-iosched: Remove prio_change logic for workload selection Wenji Huang (1): perf kmem: Fix statistics typo Wey-Yi Guy (1): iwlwifi: fix syslog message for event log dump size Williams, Mitch A (1): igbvf: Make igbvf error message more informative Wolfgang Denk (1): ARM: MX3: make CPU revision number detection work on all boards Wolfram Sang (1): i2c/pca: Don't use *_interruptible Wu Fengguang (3): ALSA: hda - HDMI sticky stream tag support memory-hotplug: add 0x prefix to HEX block_size_bytes sysdev: fix prototype for memory_sysdev_class show/store functions Wu Zhangjin (1): MIPS: Cleanup and Fixup of compressed kernel support Xiaotian Feng (2): sunrpc: fix peername failed on closed listener ACPI: don't cond_resched if irq is disabled Yan, Zheng (2): Btrfs: align offsets for btrfs_ordered_update_i_size Btrfs: Fix race in btrfs_mark_extent_written Yinghai Lu (2): x86: Fix size for ex trampoline with 32bit x86/pci: Intel ioh bus num reg accessing fix Yoichi Yuasa (13): MIPS: VR41xx: Use strlcat() for the command line arguments MIPS: AR7: Remove kgdb_enabled MIPS: PowerTV: Remove unused prom_getcmdline() MIPS: PowerTV: Remove unused ptv_memsize MIPS: PowerTV: Remove mips_machine_halt() MIPS: PowerTV: Remove unused platform_die() MIPS: PowerTV: simplify prom_init_cmdline() and merge into prom_init() MIPS: Cobalt use strlcat() for the command line arguments MIPS: AR7: Remove unused prom_getchar() MIPS: BCM63xx: Remove duplicate CONFIG_CMDLINE. MIPS: Malta, PowerTV: Remove unnecessary "Linux started" MIPS: Move vmlinux.ecoff to arch/mips/boot MIPS: Ignore vmlinux.* Yong Wang (1): hwmon: (coretemp) Fix TjMax for Atom N450/D410/D510 CPUs Yong Zhang (1): sched: Reassign prev and switch_count when reacquire_kernel_lock() fail Yongseok Koh (1): vmalloc: remove BUG_ON due to racy counting of VM_LAZY_FREE Youquan,Song (2): PCI: AER: fix aer inject result in kernel oops PCIe AER: prevent AER injection if hardware masks error reporting Zhang Rui (4): ACPI video: no warning message if "acpi_backlight=vendor" is used ACPI video: correct error-handling code ACPI: introduce kernel parameter acpi_sleep=sci_force_enable ACPI video: Prune dupe video devices, unless "video.allow_duplicates" Zhao Yakui (14): drm/i915: Add MALATA PC-81005 to ACPI LID quirk list drm/i915: Update LVDS connector status when receiving ACPI LID event drm/i915: Enable/disable the dithering for LVDS based on VBT setting drm/i915: Make the BPC in FDI rx/transcoder be consistent with that in pipeconf on Ironlake drm/i915: Select the correct BPC for LVDS on Ironlake drm/i915: Add DP dpll limit on ironlake and use existing DPLL search function drm/i915: Add HP nx9020/SamsungSX20S to ACPI LID quirk list drm/i915: Use find_pll function to calculate DPLL setting for LVDS downclock drm/i915: Fix Ironlake M/N/P ranges to match the spec drm/i915: Don't use the child device parsed from VBT to setup HDMI/DP drm/i915: Read the response after issuing DDC bus switch command drm/i915: try another possible DDC bus for the SDVO device with multiple outputs drm/i915: Fix the incorrect cursor A bit definition in DSPFW2 register ACPI: Advertise to BIOS in _OSC: _OST on _PPC changes Zhenyu Wang (10): drm/i915: implement new pm ops for i915 drm/i915: Reload hangcheck timer too for Ironlake drm/i915: remove full registers dump debug drm: remove address mask param for drm_pci_alloc() drm/i915: disable TV hotplug status check drm/i915: Don't wait interruptible for possible plane buffer flush drm/i915: parse eDP panel color depth from VBT block drm/i915: fix pixel color depth setting on eDP drm/i915: fix eDP pipe mask drm/i915: enable 36bit physical address for hardware status page Zhu Yi (4): iwlwifi: allocated rx page accounting cleanup iwl3945: fix panic in iwl3945 driver iwmc3200wifi: fix array out-of-boundary access iwlwifi: fix iwl_queue_used bug when read_ptr == write_ptr Zou Nan hai (1): drm/i915: remove loop in Ironlake interrupt handler akpm(a)linux-foundation.org (1): drivers/net/wireless/iwlwifi/iwl-tx.c: fix gcc-3.4.5 warning roel kluin (5): usbnet: test off by one atarilance: timeout ignored in lance_open() niu: timeout ignored in tcam_wait_bit() net: Test off by one in sh_eth_reset() broadcom: Fix &&/|| confusion in bcm54xx_adjust_rxrefclk() .gitignore | 1 + Documentation/DocBook/mtdnand.tmpl | 12 +- Documentation/IO-mapping.txt | 2 +- .../{DMA-mapping.txt => PCI/PCI-DMA-mapping.txt} | 0 Documentation/block/00-INDEX | 2 - Documentation/block/as-iosched.txt | 172 -- Documentation/block/biodoc.txt | 2 +- Documentation/feature-removal-schedule.txt | 49 + Documentation/filesystems/ext4.txt | 2 +- Documentation/filesystems/nilfs2.txt | 2 +- Documentation/filesystems/proc.txt | 2 - Documentation/hwmon/amc6821 | 102 ++ Documentation/hwmon/k10temp | 17 +- Documentation/ioctl/ioctl-number.txt | 203 ++- Documentation/kernel-doc-nano-HOWTO.txt | 12 +- Documentation/kernel-parameters.txt | 5 +- Documentation/kvm/api.txt | 10 +- Documentation/laptops/thinkpad-acpi.txt | 58 +- Documentation/networking/3c509.txt | 12 +- Documentation/sound/alsa/Procfile.txt | 2 +- Documentation/trace/ftrace-design.txt | 14 +- Documentation/trace/mmiotrace.txt | 15 +- Documentation/trace/ring-buffer-design.txt | 56 +- Documentation/trace/tracepoint-analysis.txt | 60 +- Documentation/vgaarbiter.txt | 2 +- MAINTAINERS | 44 +- Makefile | 5 +- arch/alpha/include/asm/topology.h | 3 + arch/arm/Kconfig | 73 +- arch/arm/Makefile | 4 +- arch/arm/boot/compressed/Makefile | 34 +- arch/arm/boot/compressed/head.S | 8 + arch/arm/boot/compressed/misc.c | 129 +- arch/arm/boot/compressed/{piggy.S => piggy.gzip.S} | 2 +- arch/arm/boot/compressed/{piggy.S => piggy.lzo.S} | 2 +- arch/arm/configs/u300_defconfig | 81 +- arch/arm/include/asm/cacheflush.h | 20 +- arch/arm/include/asm/cpu.h | 1 + arch/arm/include/asm/dma.h | 4 +- arch/arm/include/asm/ptrace.h | 6 + arch/arm/include/asm/unistd.h | 1 + arch/arm/kernel/debug.S | 20 + arch/arm/kernel/elf.c | 9 - arch/arm/kernel/entry-armv.S | 4 +- arch/arm/kernel/process.c | 3 +- arch/arm/mach-davinci/board-dm355-evm.c | 1 + arch/arm/mach-davinci/board-dm365-evm.c | 11 +- arch/arm/mach-davinci/board-dm644x-evm.c | 1 + arch/arm/mach-davinci/cp_intc.c | 11 + arch/arm/mach-davinci/devices-da8xx.c | 9 +- arch/arm/mach-davinci/dm365.c | 1 - arch/arm/mach-davinci/include/mach/keyscan.h | 1 + arch/arm/mach-gemini/include/mach/uncompress.h | 4 +- arch/arm/mach-lh7a40x/clocks.c | 2 + arch/arm/mach-mmp/ttc_dkb.c | 2 - arch/arm/mach-mx2/mxt_td60.c | 24 - arch/arm/mach-mx25/clock.c | 2 + arch/arm/mach-mx25/devices.c | 19 + arch/arm/mach-mx25/devices.h | 1 + arch/arm/mach-mx25/mx25pdk.c | 46 +- arch/arm/mach-mx3/Kconfig | 3 +- arch/arm/mach-mx3/mm.c | 5 + arch/arm/mach-mx3/mx31ads.c | 5 - arch/arm/mach-mx3/mx31lite.c | 9 +- arch/arm/mach-mx3/mx31moboard-devboard.c | 2 +- arch/arm/mach-mx3/mx31moboard-marxbot.c | 2 +- arch/arm/mach-mx3/mx31moboard.c | 7 +- arch/arm/mach-mx3/mx31pdk.c | 5 - arch/arm/mach-mx3/pcm037.c | 32 +- arch/arm/mach-omap1/clock_data.c | 10 +- arch/arm/mach-omap1/devices.c | 35 + arch/arm/mach-omap1/mux.c | 8 + arch/arm/mach-omap2/Kconfig | 1 + arch/arm/mach-omap2/board-zoom-peripherals.c | 8 +- arch/arm/mach-omap2/clock2xxx.c | 57 +- arch/arm/mach-omap2/clock34xx.c | 1 - arch/arm/mach-omap2/clock34xx_data.c | 6 +- arch/arm/mach-omap2/clockdomain.c | 4 +- arch/arm/mach-omap2/io.c | 19 +- arch/arm/mach-omap2/mux.c | 6 +- arch/arm/mach-omap2/opp2420_data.c | 38 +- arch/arm/mach-omap2/opp2430_data.c | 30 +- arch/arm/mach-omap2/pm34xx.c | 4 +- arch/arm/mach-omap2/serial.c | 12 +- arch/arm/mach-pxa/Kconfig | 2 + arch/arm/mach-pxa/include/mach/hardware.h | 9 +- arch/arm/mach-pxa/include/mach/zylonite.h | 7 - arch/arm/mach-pxa/littleton.c | 6 + arch/arm/mach-pxa/magician.c | 2 +- arch/arm/mach-pxa/palmld.c | 2 +- arch/arm/mach-pxa/palmt5.c | 2 +- arch/arm/mach-pxa/palmtc.c | 2 +- arch/arm/mach-pxa/palmte2.c | 2 +- arch/arm/mach-pxa/palmtreo.c | 2 +- arch/arm/mach-pxa/palmtx.c | 2 +- arch/arm/mach-pxa/palmz72.c | 2 +- arch/arm/mach-pxa/poodle.c | 2 +- arch/arm/mach-pxa/spitz.c | 4 +- arch/arm/mach-pxa/viper.c | 2 +- arch/arm/mach-pxa/zeus.c | 36 +- arch/arm/mach-pxa/zylonite.c | 87 +- arch/arm/mach-pxa/zylonite_pxa300.c | 12 +- arch/arm/mach-pxa/zylonite_pxa320.c | 4 - arch/arm/mach-realview/include/mach/board-pb1176.h | 4 +- arch/arm/mach-realview/include/mach/platform.h | 2 +- arch/arm/mach-realview/realview_eb.c | 15 + arch/arm/mach-realview/realview_pb1176.c | 10 +- arch/arm/mach-realview/realview_pb11mp.c | 11 +- arch/arm/mach-realview/realview_pba8.c | 15 + arch/arm/mach-realview/realview_pbx.c | 15 + arch/arm/mach-s3c2410/include/mach/ts.h | 21 + arch/arm/mach-s3c2410/mach-bast.c | 3 + arch/arm/mach-s3c2410/mach-h1940.c | 8 + arch/arm/mach-s3c2440/mach-mini2440.c | 36 +- arch/arm/mach-s3c2440/mach-osiris.c | 2 + arch/arm/mach-s3c6410/mach-smdk6410.c | 1 + arch/arm/mach-w90x900/include/mach/system.h | 15 +- arch/arm/mach-w90x900/time.c | 64 +- arch/arm/mm/Makefile | 12 + arch/arm/mm/cache-xsc3l2.c | 11 +- arch/arm/mm/init.c | 2 +- arch/arm/mm/mmu.c | 2 + arch/arm/mm/proc-v6.S | 2 - arch/arm/mm/proc-v7.S | 11 +- arch/arm/mm/proc-xsc3.S | 7 + arch/arm/mm/tlb-v7.S | 1 - arch/arm/plat-mxc/include/mach/iomux-mx25.h | 58 +- arch/arm/plat-mxc/include/mach/mx25.h | 4 + arch/arm/plat-mxc/include/mach/uncompress.h | 4 +- arch/arm/plat-omap/clock.c | 16 +- arch/arm/plat-omap/cpu-omap.c | 1 + arch/arm/plat-omap/gpio.c | 63 +- arch/arm/plat-omap/include/plat/board.h | 1 - arch/arm/plat-omap/include/plat/clock.h | 2 + arch/arm/plat-omap/include/plat/control.h | 2 +- arch/arm/plat-omap/include/plat/io.h | 36 +- arch/arm/plat-omap/include/plat/mux.h | 8 + arch/arm/plat-omap/include/plat/omap7xx.h | 3 + arch/arm/plat-omap/io.c | 12 +- arch/arm/plat-omap/iommu.c | 2 +- arch/arm/plat-omap/mcbsp.c | 20 +- arch/arm/plat-pxa/pwm.c | 8 +- arch/arm/plat-s3c/dev-nand.c | 8 +- arch/arm/plat-s3c/include/plat/devs.h | 1 + arch/arm/plat-s3c24xx/devs.c | 18 + arch/arm/plat-s3c24xx/s3c244x.c | 1 + arch/arm/plat-s3c64xx/s3c6400-clock.c | 10 +- arch/blackfin/include/asm/page.h | 5 + arch/blackfin/kernel/kgdb.c | 207 +--- arch/blackfin/mm/Makefile | 2 +- arch/blackfin/mm/maccess.c | 97 + arch/frv/include/asm/page.h | 2 - arch/ia64/include/asm/ftrace.h | 1 - arch/ia64/include/asm/kprobes.h | 5 - arch/ia64/include/asm/tlb.h | 2 +- arch/ia64/include/asm/topology.h | 4 +- arch/ia64/include/asm/types.h | 5 + arch/ia64/kernel/mca.c | 5 +- arch/ia64/kernel/perfmon.c | 2 +- arch/ia64/kvm/vcpu.h | 9 +- arch/ia64/kvm/vmm.c | 4 +- arch/ia64/kvm/vtlb.c | 2 +- arch/ia64/mm/init.c | 2 +- arch/ia64/mm/tlb.c | 32 +- arch/m68k/include/asm/io_no.h | 14 +- arch/m68k/include/asm/page_no.h | 4 +- arch/m68k/include/asm/virtconvert.h | 4 + arch/m68knommu/platform/coldfire/pit.c | 1 - arch/microblaze/Kconfig | 2 +- arch/microblaze/include/asm/ptrace.h | 14 + arch/microblaze/include/asm/unistd.h | 5 +- arch/microblaze/kernel/prom_parse.c | 2 +- arch/microblaze/kernel/syscall_table.S | 2 +- arch/mips/alchemy/common/dbdma.c | 4 +- arch/mips/ar7/prom.c | 15 - arch/mips/bcm63xx/boards/board_bcm963xx.c | 33 +- arch/mips/bcm63xx/prom.c | 3 - arch/mips/boot/.gitignore | 1 + arch/mips/boot/Makefile | 2 +- arch/mips/boot/compressed/Makefile | 9 +- arch/mips/boot/compressed/decompress.c | 10 - arch/mips/boot/compressed/ld.script | 195 +-- arch/mips/cavium-octeon/csrc-octeon.c | 32 + arch/mips/cobalt/setup.c | 24 +- arch/mips/include/asm/mach-ip27/topology.h | 4 +- arch/mips/mipssim/sim_setup.c | 4 - arch/mips/mm/init.c | 2 +- arch/mips/mm/tlbex.c | 8 - arch/mips/mti-malta/malta-init.c | 1 - arch/mips/powertv/Makefile | 2 +- arch/mips/powertv/cmdline.c | 52 - arch/mips/powertv/init.c | 15 +- arch/mips/powertv/init.h | 2 - arch/mips/powertv/memory.c | 5 - arch/mips/powertv/powertv_setup.c | 21 - arch/mips/powertv/reset.c | 18 - arch/mips/powertv/time.c | 1 - arch/mips/sgi-ip27/ip27-memory.c | 2 +- arch/mips/txx9/generic/setup.c | 21 - arch/mips/vr41xx/common/init.c | 6 +- arch/mn10300/Makefile | 2 +- arch/mn10300/configs/asb2303_defconfig | 107 +- arch/mn10300/include/asm/bitops.h | 4 +- arch/mn10300/include/asm/div64.h | 2 + arch/mn10300/include/asm/system.h | 1 + arch/mn10300/include/asm/tlbflush.h | 2 +- arch/mn10300/include/asm/uaccess.h | 4 +- arch/mn10300/include/asm/unistd.h | 3 +- arch/mn10300/kernel/entry.S | 1 + arch/mn10300/kernel/mn10300-serial.c | 3 +- arch/mn10300/kernel/signal.c | 2 +- arch/mn10300/lib/checksum.c | 1 + arch/mn10300/lib/delay.c | 3 +- arch/mn10300/lib/usercopy.c | 6 +- arch/mn10300/mm/dma-alloc.c | 17 + arch/mn10300/mm/init.c | 3 +- arch/mn10300/mm/misalignment.c | 4 +- arch/mn10300/unit-asb2305/include/unit/serial.h | 2 +- arch/mn10300/unit-asb2305/include/unit/timex.h | 2 +- arch/mn10300/unit-asb2305/leds.c | 4 +- arch/mn10300/unit-asb2305/pci-asb2305.c | 39 - arch/mn10300/unit-asb2305/pci-asb2305.h | 1 - arch/mn10300/unit-asb2305/pci.c | 62 +- arch/mn10300/unit-asb2305/unit-init.c | 5 +- arch/powerpc/boot/wrapper | 7 +- arch/powerpc/configs/52xx/cm5200_defconfig | 75 +- arch/powerpc/configs/52xx/lite5200b_defconfig | 90 +- arch/powerpc/configs/52xx/motionpro_defconfig | 79 +- arch/powerpc/configs/52xx/pcm030_defconfig | 83 +- arch/powerpc/configs/52xx/tqm5200_defconfig | 79 +- arch/powerpc/configs/83xx/asp8347_defconfig | 88 +- arch/powerpc/configs/83xx/kmeter1_defconfig | 68 +- arch/powerpc/configs/83xx/mpc8313_rdb_defconfig | 97 +- arch/powerpc/configs/83xx/mpc8315_rdb_defconfig | 100 +- arch/powerpc/configs/83xx/mpc832x_mds_defconfig | 89 +- arch/powerpc/configs/83xx/mpc832x_rdb_defconfig | 93 +- arch/powerpc/configs/83xx/mpc834x_itx_defconfig | 95 +- arch/powerpc/configs/83xx/mpc834x_itxgp_defconfig | 92 +- arch/powerpc/configs/83xx/mpc834x_mds_defconfig | 86 +- arch/powerpc/configs/83xx/mpc836x_mds_defconfig | 89 +- arch/powerpc/configs/83xx/mpc836x_rdk_defconfig | 87 +- arch/powerpc/configs/83xx/mpc837x_mds_defconfig | 89 +- arch/powerpc/configs/83xx/mpc837x_rdb_defconfig | 91 +- arch/powerpc/configs/83xx/sbc834x_defconfig | 85 +- arch/powerpc/configs/85xx/ksi8560_defconfig | 69 +- arch/powerpc/configs/85xx/mpc8540_ads_defconfig | 69 +- arch/powerpc/configs/85xx/mpc8560_ads_defconfig | 76 +- arch/powerpc/configs/85xx/mpc85xx_cds_defconfig | 75 +- arch/powerpc/configs/85xx/sbc8548_defconfig | 77 +- arch/powerpc/configs/85xx/sbc8560_defconfig | 72 +- arch/powerpc/configs/85xx/socrates_defconfig | 99 +- arch/powerpc/configs/85xx/stx_gp3_defconfig | 87 +- arch/powerpc/configs/85xx/tqm8540_defconfig | 89 +- arch/powerpc/configs/85xx/tqm8541_defconfig | 90 +- arch/powerpc/configs/85xx/tqm8548_defconfig | 95 +- arch/powerpc/configs/85xx/tqm8555_defconfig | 90 +- arch/powerpc/configs/85xx/tqm8560_defconfig | 90 +- arch/powerpc/configs/85xx/xes_mpc85xx_defconfig | 103 +- arch/powerpc/configs/86xx/gef_ppc9a_defconfig | 102 +- arch/powerpc/configs/86xx/gef_sbc310_defconfig | 99 +- arch/powerpc/configs/86xx/gef_sbc610_defconfig | 92 +- arch/powerpc/configs/86xx/mpc8610_hpcd_defconfig | 88 +- arch/powerpc/configs/86xx/mpc8641_hpcn_defconfig | 90 +- arch/powerpc/configs/86xx/sbc8641d_defconfig | 81 +- arch/powerpc/configs/adder875_defconfig | 57 +- arch/powerpc/configs/c2k_defconfig | 90 +- arch/powerpc/configs/ep8248e_defconfig | 64 +- arch/powerpc/configs/ep88xc_defconfig | 55 +- arch/powerpc/configs/linkstation_defconfig | 96 +- arch/powerpc/configs/mgcoge_defconfig | 74 +- arch/powerpc/configs/mgsuvd_defconfig | 63 +- arch/powerpc/configs/mpc5200_defconfig | 106 +- arch/powerpc/configs/mpc7448_hpc2_defconfig | 79 +- arch/powerpc/configs/mpc8272_ads_defconfig | 70 +- arch/powerpc/configs/mpc83xx_defconfig | 92 +- arch/powerpc/configs/mpc85xx_defconfig | 97 +- arch/powerpc/configs/mpc85xx_smp_defconfig | 97 +- arch/powerpc/configs/mpc866_ads_defconfig | 65 +- arch/powerpc/configs/mpc86xx_defconfig | 93 +- arch/powerpc/configs/mpc885_ads_defconfig | 55 +- arch/powerpc/configs/pq2fads_defconfig | 72 +- arch/powerpc/configs/prpmc2800_defconfig | 94 +- arch/powerpc/configs/storcenter_defconfig | 86 +- arch/powerpc/include/asm/topology.h | 4 +- arch/powerpc/kernel/head_8xx.S | 12 +- arch/powerpc/kernel/pci-common.c | 13 + arch/powerpc/kernel/rtas-proc.c | 14 +- arch/powerpc/kernel/smp.c | 12 + arch/powerpc/kernel/swsusp_32.S | 2 +- arch/powerpc/kernel/time.c | 11 +- arch/powerpc/kvm/book3s_64_mmu.c | 22 +- arch/powerpc/mm/mmap_64.c | 4 +- arch/powerpc/platforms/cell/spufs/coredump.c | 2 +- arch/powerpc/platforms/iseries/vio.c | 2 + arch/powerpc/platforms/pseries/dlpar.c | 18 +- arch/powerpc/platforms/pseries/xics.c | 12 +- arch/s390/defconfig | 86 +- arch/s390/include/asm/bug.h | 4 + arch/s390/include/asm/param.h | 28 +- arch/s390/include/asm/sigp.h | 20 +- arch/s390/include/asm/thread_info.h | 12 +- arch/s390/include/asm/unistd.h | 6 +- arch/s390/kernel/compat_linux.c | 41 +- arch/s390/kernel/compat_wrapper.S | 9 - arch/s390/kernel/process.c | 3 +- arch/s390/kernel/ptrace.c | 8 + arch/s390/kernel/setup.c | 8 - arch/s390/kernel/signal.c | 10 +- arch/s390/kernel/smp.c | 9 + arch/s390/kernel/syscalls.S | 1 - arch/s390/kernel/topology.c | 2 +- arch/s390/lib/spinlock.c | 2 +- arch/s390/mm/mmap.c | 4 +- arch/score/mm/init.c | 2 +- arch/sh/Kconfig | 16 +- arch/sh/boards/mach-ecovec24/setup.c | 8 + arch/sh/boards/mach-se/7724/setup.c | 2 +- arch/sh/configs/ap325rxa_defconfig | 87 +- arch/sh/configs/cayman_defconfig | 168 ++- arch/sh/configs/dreamcast_defconfig | 78 +- arch/sh/configs/ecovec24-romimage_defconfig | 76 +- arch/sh/configs/ecovec24_defconfig | 89 +- arch/sh/configs/edosk7705_defconfig | 49 +- arch/sh/configs/edosk7760_defconfig | 70 +- arch/sh/configs/espt_defconfig | 77 +- arch/sh/configs/hp6xx_defconfig | 70 +- arch/sh/configs/kfr2r09-romimage_defconfig | 58 +- arch/sh/configs/kfr2r09_defconfig | 71 +- arch/sh/configs/landisk_defconfig | 85 +- arch/sh/configs/lboxre2_defconfig | 89 +- arch/sh/configs/magicpanelr2_defconfig | 69 +- arch/sh/configs/microdev_defconfig | 71 +- arch/sh/configs/migor_defconfig | 92 +- arch/sh/configs/polaris_defconfig | 77 +- arch/sh/configs/r7780mp_defconfig | 99 +- arch/sh/configs/r7785rp_defconfig | 98 +- arch/sh/configs/rsk7201_defconfig | 59 +- arch/sh/configs/rsk7203_defconfig | 71 +- arch/sh/configs/rts7751r2d1_defconfig | 95 +- arch/sh/configs/rts7751r2dplus_defconfig | 95 +- arch/sh/configs/sdk7780_defconfig | 88 +- arch/sh/configs/se7206_defconfig | 66 +- arch/sh/configs/se7343_defconfig | 72 +- arch/sh/configs/se7619_defconfig | 58 +- arch/sh/configs/se7705_defconfig | 71 +- arch/sh/configs/se7712_defconfig | 61 +- arch/sh/configs/se7721_defconfig | 62 +- arch/sh/configs/se7722_defconfig | 69 +- arch/sh/configs/se7724_defconfig | 100 +- arch/sh/configs/se7750_defconfig | 70 +- arch/sh/configs/se7751_defconfig | 70 +- arch/sh/configs/se7780_defconfig | 79 +- arch/sh/configs/sh03_defconfig | 80 +- arch/sh/configs/sh7710voipgw_defconfig | 66 +- arch/sh/configs/sh7724_generic_defconfig | 80 +- arch/sh/configs/sh7763rdp_defconfig | 74 +- arch/sh/configs/sh7770_generic_defconfig | 79 +- arch/sh/configs/sh7785lcr_32bit_defconfig | 83 +- arch/sh/configs/sh7785lcr_defconfig | 83 +- arch/sh/configs/shmin_defconfig | 64 +- arch/sh/configs/shx3_defconfig | 93 +- arch/sh/configs/snapgear_defconfig | 79 +- arch/sh/configs/systemh_defconfig | 64 +- arch/sh/configs/titan_defconfig | 87 +- arch/sh/configs/ul2_defconfig | 86 +- arch/sh/configs/urquell_defconfig | 99 +- arch/sh/include/asm/siu.h | 26 + arch/sh/include/asm/timex.h | 10 + arch/sh/include/asm/unistd_32.h | 5 +- arch/sh/include/asm/unistd_64.h | 3 +- arch/sh/kernel/syscalls_32.S | 1 - arch/sh/kernel/syscalls_64.S | 1 + arch/sh/mm/Kconfig | 7 +- arch/sh/tools/Makefile | 2 +- arch/sparc/kernel/nmi.c | 3 +- arch/sparc/kernel/perf_event.c | 11 + arch/um/drivers/mconsole_kern.c | 49 +- arch/x86/Kconfig | 6 + arch/x86/Kconfig.cpu | 2 +- arch/x86/boot/compressed/Makefile | 6 +- arch/x86/boot/compressed/misc.c | 4 + arch/x86/include/asm/kvm.h | 4 + arch/x86/include/asm/mce.h | 3 +- arch/x86/include/asm/perf_event.h | 1 + arch/x86/include/asm/uaccess_32.h | 5 +- arch/x86/include/asm/uaccess_64.h | 5 +- arch/x86/include/asm/uv/uv_hub.h | 98 +- arch/x86/kernel/acpi/boot.c | 22 +- arch/x86/kernel/acpi/sleep.c | 2 + arch/x86/kernel/amd_iommu_init.c | 10 + arch/x86/kernel/aperture_64.c | 1 + arch/x86/kernel/apic/apic.c | 23 +- arch/x86/kernel/apic/apic_flat_64.c | 5 + arch/x86/kernel/apic/io_apic.c | 7 + arch/x86/kernel/apic/probe_64.c | 13 +- arch/x86/kernel/apic/x2apic_uv_x.c | 38 +- arch/x86/kernel/cpu/perf_event.c | 9 +- arch/x86/kernel/dumpstack.c | 14 +- arch/x86/kernel/e820.c | 4 +- arch/x86/kernel/process.c | 6 +- arch/x86/kernel/process_32.c | 14 +- arch/x86/kernel/process_64.c | 24 +- arch/x86/kvm/lapic.c | 1 + arch/x86/kvm/paging_tmpl.h | 18 - arch/x86/kvm/x86.c | 12 +- arch/x86/mm/init_32.c | 3 +- arch/x86/mm/kmemcheck/error.c | 19 +- arch/x86/mm/kmmio.c | 7 +- arch/x86/pci/bus_numa.c | 2 +- arch/x86/pci/intel_bus.c | 4 + arch/x86/tools/chkobjdump.awk | 16 +- arch/x86/xen/enlighten.c | 4 + block/blk-barrier.c | 2 +- block/blk-ioc.c | 5 - block/blk-settings.c | 140 ++- block/cfq-iosched.c | 73 +- block/genhd.c | 2 +- crypto/async_tx/raid6test.c | 7 + drivers/Kconfig | 2 +- drivers/acpi/acpi_pad.c | 37 +- drivers/acpi/bus.c | 7 +- drivers/acpi/ec.c | 126 +- drivers/acpi/pci_link.c | 2 +- drivers/acpi/pci_root.c | 2 +- drivers/acpi/power.c | 2 +- drivers/acpi/power_meter.c | 4 +- drivers/acpi/processor_idle.c | 64 +- drivers/acpi/processor_pdc.c | 32 +- drivers/acpi/processor_thermal.c | 3 +- drivers/acpi/sbs.c | 3 + drivers/acpi/sbshc.c | 2 +- drivers/acpi/sleep.c | 29 +- drivers/acpi/video.c | 51 +- drivers/ata/ata_piix.c | 2 +- drivers/ata/libata-core.c | 38 +- drivers/ata/libata-eh.c | 5 +- drivers/ata/sata_promise.c | 2 +- drivers/base/devtmpfs.c | 3 +- drivers/base/memory.c | 2 +- drivers/base/power/main.c | 9 +- drivers/block/DAC960.c | 2 +- drivers/block/aoe/aoecmd.c | 17 +- drivers/block/drbd/Kconfig | 2 +- drivers/block/drbd/drbd_int.h | 9 +- drivers/block/drbd/drbd_main.c | 6 +- drivers/block/drbd/drbd_nl.c | 19 +- drivers/block/drbd/drbd_proc.c | 2 +- drivers/block/drbd/drbd_receiver.c | 47 +- drivers/block/drbd/drbd_worker.c | 2 - drivers/block/mg_disk.c | 2 +- drivers/char/Kconfig | 2 +- drivers/char/agp/amd64-agp.c | 10 +- drivers/char/agp/backend.c | 13 +- drivers/char/agp/hp-agp.c | 6 +- drivers/char/agp/intel-agp.c | 6 +- drivers/char/hw_random/core.c | 5 +- drivers/char/hw_random/virtio-rng.c | 6 +- drivers/char/ipmi/ipmi_si_intf.c | 4 +- drivers/char/nozomi.c | 2 +- drivers/char/sonypi.c | 11 +- drivers/char/toshiba.c | 12 +- drivers/char/tty_io.c | 2 +- drivers/cpuidle/governors/menu.c | 12 +- drivers/dma/at_hdmac.c | 4 +- drivers/dma/coh901318.c | 2 +- drivers/dma/dw_dmac.c | 2 - drivers/dma/ioat/dma.c | 2 +- drivers/dma/ioat/dma.h | 18 + drivers/dma/ioat/dma_v2.c | 69 +- drivers/dma/ioat/dma_v2.h | 2 + drivers/dma/ioat/dma_v3.c | 60 +- drivers/dma/ioat/registers.h | 1 + drivers/dma/shdma.c | 324 +++-- drivers/dma/shdma.h | 9 +- drivers/edac/amd64_edac.c | 2 +- drivers/edac/i5000_edac.c | 8 +- drivers/firewire/Kconfig | 44 +- drivers/firewire/core-cdev.c | 27 +- drivers/firewire/core-transaction.c | 118 +- drivers/firewire/ohci.c | 4 +- drivers/gpio/Kconfig | 9 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| 11 +- drivers/isdn/hardware/mISDN/hfcmulti.c | 2 +- drivers/lguest/segments.c | 4 +- drivers/macintosh/smu.c | 2 +- drivers/macintosh/therm_pm72.c | 2 +- drivers/macintosh/therm_windtunnel.c | 2 +- drivers/md/dm-table.c | 20 +- drivers/md/md.c | 42 +- drivers/media/IR/ir-keytable.c | 2 +- drivers/media/common/tuners/tda8290.c | 12 +- drivers/media/dvb/Kconfig | 4 + drivers/media/dvb/Makefile | 14 +- drivers/media/dvb/firewire/firedtv-fw.c | 12 +- drivers/media/dvb/frontends/Kconfig | 19 + drivers/media/dvb/frontends/Makefile | 2 + drivers/media/dvb/frontends/dib8000.h | 2 +- drivers/media/dvb/frontends/lgdt3305.h | 6 +- drivers/media/dvb/frontends/mb86a16.c | 1878 ++++++++++++++++++++ drivers/media/dvb/frontends/mb86a16.h | 52 + drivers/media/dvb/frontends/mb86a16_priv.h | 151 ++ drivers/media/dvb/frontends/tda10021.c | 4 + drivers/media/dvb/frontends/tda665x.c | 257 +++ drivers/media/dvb/frontends/tda665x.h | 52 + drivers/media/dvb/mantis/Kconfig | 32 + drivers/media/dvb/mantis/Makefile | 28 + drivers/media/dvb/mantis/hopper_cards.c | 275 +++ drivers/media/dvb/mantis/hopper_vp3028.c | 88 + drivers/media/dvb/mantis/hopper_vp3028.h | 30 + drivers/media/dvb/mantis/mantis_ca.c | 207 +++ drivers/media/dvb/mantis/mantis_ca.h | 27 + drivers/media/dvb/mantis/mantis_cards.c | 305 ++++ drivers/media/dvb/mantis/mantis_common.h | 179 ++ drivers/media/dvb/mantis/mantis_core.c | 238 +++ drivers/media/dvb/mantis/mantis_core.h | 57 + drivers/media/dvb/mantis/mantis_dma.c | 256 +++ drivers/media/dvb/mantis/mantis_dma.h | 30 + drivers/media/dvb/mantis/mantis_dvb.c | 296 +++ drivers/media/dvb/mantis/mantis_dvb.h | 35 + drivers/media/dvb/mantis/mantis_evm.c | 117 ++ drivers/media/dvb/mantis/mantis_hif.c | 240 +++ drivers/media/dvb/mantis/mantis_hif.h | 29 + drivers/media/dvb/mantis/mantis_i2c.c | 267 +++ drivers/media/dvb/mantis/mantis_i2c.h | 30 + drivers/media/dvb/mantis/mantis_input.c | 148 ++ drivers/media/dvb/mantis/mantis_ioc.c | 130 ++ drivers/media/dvb/mantis/mantis_ioc.h | 51 + drivers/media/dvb/mantis/mantis_link.h | 83 + drivers/media/dvb/mantis/mantis_pci.c | 177 ++ drivers/media/dvb/mantis/mantis_pci.h | 27 + drivers/media/dvb/mantis/mantis_pcmcia.c | 120 ++ drivers/media/dvb/mantis/mantis_reg.h | 197 ++ drivers/media/dvb/mantis/mantis_uart.c | 186 ++ drivers/media/dvb/mantis/mantis_uart.h | 58 + drivers/media/dvb/mantis/mantis_vp1033.c | 212 +++ drivers/media/dvb/mantis/mantis_vp1033.h | 30 + drivers/media/dvb/mantis/mantis_vp1034.c | 119 ++ drivers/media/dvb/mantis/mantis_vp1034.h | 33 + drivers/media/dvb/mantis/mantis_vp1041.c | 358 ++++ drivers/media/dvb/mantis/mantis_vp1041.h | 33 + drivers/media/dvb/mantis/mantis_vp2033.c | 187 ++ drivers/media/dvb/mantis/mantis_vp2033.h | 30 + drivers/media/dvb/mantis/mantis_vp2040.c | 186 ++ drivers/media/dvb/mantis/mantis_vp2040.h | 32 + drivers/media/dvb/mantis/mantis_vp3028.c | 38 + drivers/media/dvb/mantis/mantis_vp3028.h | 33 + 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| 119 +- drivers/mfd/tc6393xb.c | 56 +- drivers/mfd/tmio_core.c | 52 + drivers/mfd/twl4030-irq.c | 4 +- drivers/mfd/wm8350-core.c | 3 +- drivers/mfd/wm8350-irq.c | 4 +- drivers/mmc/card/block.c | 8 + drivers/mmc/card/queue.c | 18 +- drivers/mmc/core/mmc.c | 2 +- drivers/mmc/host/tmio_mmc.c | 59 +- drivers/mmc/host/tmio_mmc.h | 46 +- drivers/net/3c507.c | 4 +- drivers/net/Kconfig | 2 + drivers/net/arm/Kconfig | 1 + drivers/net/atarilance.c | 2 +- drivers/net/atlx/atl2.c | 7 +- drivers/net/benet/be.h | 1 + drivers/net/benet/be_cmds.c | 36 + drivers/net/benet/be_cmds.h | 16 + drivers/net/benet/be_ethtool.c | 77 +- drivers/net/bnx2x_main.c | 2 + drivers/net/bonding/bond_3ad.c | 2 +- drivers/net/can/mcp251x.c | 2 +- drivers/net/cs89x0.c | 3 +- drivers/net/davinci_emac.c | 4 +- drivers/net/e1000e/82571.c | 2 + drivers/net/e1000e/e1000.h | 1 - drivers/net/e1000e/es2lan.c | 2 + drivers/net/e1000e/hw.h | 1 + drivers/net/e1000e/ich8lan.c | 78 +- drivers/net/e1000e/lib.c | 54 +- 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+++- drivers/net/sfc/selftest.c | 8 +- drivers/net/sfc/siena.c | 1 + drivers/net/sfc/tenxpress.c | 138 +- drivers/net/sfc/tx.c | 4 +- drivers/net/sh_eth.c | 2 +- drivers/net/sky2.c | 4 +- drivers/net/tg3.c | 27 +- drivers/net/tg3.h | 3 + drivers/net/tulip/Kconfig | 4 + drivers/net/tulip/dmfe.c | 21 + drivers/net/tulip/tulip_core.c | 32 +- drivers/net/tun.c | 6 +- drivers/net/ucc_geth.c | 45 +- drivers/net/ucc_geth.h | 13 +- drivers/net/usb/hso.c | 105 +- drivers/net/usb/rtl8150.c | 4 +- drivers/net/via-rhine.c | 41 +- drivers/net/via-velocity.c | 8 +- drivers/net/vxge/vxge-main.c | 8 +- drivers/net/wireless/ath/ath5k/base.c | 18 +- drivers/net/wireless/ath/ath5k/eeprom.c | 32 +- drivers/net/wireless/ath/ath5k/eeprom.h | 8 + drivers/net/wireless/ath/ath9k/Kconfig | 2 +- drivers/net/wireless/ath/ath9k/ath9k.h | 6 +- drivers/net/wireless/ath/ath9k/mac.c | 2 +- drivers/net/wireless/ath/ath9k/mac.h | 3 + drivers/net/wireless/ath/ath9k/main.c | 20 + drivers/net/wireless/ath/ath9k/pci.c | 2 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drivers/net/wireless/zd1211rw/zd_mac.c | 16 +- drivers/pci/hotplug/shpchp.h | 2 +- drivers/pci/intel-iommu.c | 6 +- drivers/pci/intr_remapping.c | 2 +- drivers/pci/pci-acpi.c | 10 +- drivers/pci/pci-sysfs.c | 6 +- drivers/pci/pci.c | 34 +- drivers/pci/pci.h | 8 + drivers/pci/pcie/aer/Kconfig.debug | 4 +- drivers/pci/pcie/aer/aer_inject.c | 34 +- drivers/pci/pcie/aer/aerdrv.c | 2 +- drivers/pci/pcie/aer/aerdrv_acpi.c | 2 +- drivers/pci/pcie/aer/aerdrv_core.c | 6 +- drivers/pci/pcie/aer/aerdrv_errprint.c | 4 +- drivers/pci/pcie/aspm.c | 4 +- drivers/pci/pcie/portdrv_core.c | 16 +- drivers/pci/pcie/portdrv_pci.c | 19 +- drivers/pci/quirks.c | 74 +- drivers/pci/search.c | 6 +- drivers/pcmcia/cardbus.c | 2 +- drivers/platform/x86/Kconfig | 29 + drivers/platform/x86/dell-wmi.c | 18 +- drivers/platform/x86/eeepc-laptop.c | 298 ++-- drivers/platform/x86/hp-wmi.c | 11 +- drivers/platform/x86/msi-wmi.c | 9 +- drivers/platform/x86/sony-laptop.c | 9 +- drivers/platform/x86/thinkpad_acpi.c | 59 +- 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drivers/staging/et131x/et1310_rx.c | 6 +- drivers/staging/hv/Hv.c | 50 +- drivers/staging/hv/Hv.h | 6 +- drivers/staging/hv/Vmbus.c | 12 +- drivers/usb/core/devices.c | 2 +- drivers/usb/core/hcd.c | 18 + drivers/usb/core/hub.c | 18 +- drivers/usb/core/message.c | 8 +- drivers/usb/core/sysfs.c | 6 + drivers/usb/host/ehci-hcd.c | 5 +- drivers/usb/host/ehci-hub.c | 20 +- drivers/usb/host/ehci-q.c | 11 +- drivers/usb/host/fhci-hcd.c | 3 +- drivers/usb/host/isp1362-hcd.c | 25 +- drivers/usb/host/isp1760-hcd.c | 6 +- drivers/usb/host/uhci-hcd.c | 15 +- drivers/usb/host/uhci-hub.c | 2 +- drivers/usb/serial/generic.c | 10 +- drivers/usb/storage/unusual_devs.h | 7 - drivers/usb/storage/usb.c | 3 +- drivers/video/backlight/omap1_bl.c | 2 - drivers/video/omap/dispc.c | 18 +- drivers/video/omap/lcd_htcherald.c | 4 +- drivers/video/omap/omapfb.h | 2 + drivers/video/omap/omapfb_main.c | 25 + drivers/video/omap/rfbi.c | 4 +- drivers/video/omap2/dss/Kconfig | 7 + drivers/video/omap2/dss/core.c | 10 + drivers/video/omap2/dss/dispc.c | 74 +- drivers/video/omap2/dss/dsi.c | 159 ++- drivers/video/omap2/dss/dss.c | 6 +- drivers/video/omap2/dss/dss.h | 14 + drivers/video/omap2/dss/rfbi.c | 30 +- drivers/video/omap2/omapfb/omapfb-main.c | 6 + drivers/video/pxafb.c | 4 +- drivers/video/s3c-fb.c | 14 +- drivers/video/via/accel.c | 5 +- drivers/video/via/viafbdev.c | 11 +- drivers/virtio/virtio_balloon.c | 6 +- drivers/watchdog/iTCO_wdt.c | 45 +- drivers/xen/manage.c | 8 +- firmware/Makefile | 3 +- firmware/WHENCE | 1 + firmware/cis/PE520.cis.ihex | 9 + fs/9p/vfs_inode.c | 41 +- fs/binfmt_elf_fdpic.c | 17 +- fs/bio.c | 2 +- fs/btrfs/acl.c | 12 +- fs/btrfs/extent-tree.c | 32 +- fs/btrfs/file.c | 100 +- fs/btrfs/inode.c | 12 +- fs/btrfs/ordered-data.c | 2 + fs/btrfs/relocation.c | 4 +- fs/btrfs/volumes.c | 4 +- fs/cifs/CHANGES | 4 +- fs/cifs/cifs_dfs_ref.c | 3 +- fs/cifs/cifsfs.c | 3 +- fs/cifs/connect.c | 13 +- fs/compat_ioctl.c | 3 + fs/configfs/symlink.c | 4 +- fs/ecryptfs/crypto.c | 4 +- fs/ecryptfs/file.c | 17 +- fs/ecryptfs/inode.c | 158 ++- fs/ecryptfs/main.c | 4 +- fs/exofs/inode.c | 17 +- fs/exofs/pnfs.h | 10 +- fs/ext4/Kconfig | 1 + fs/ext4/block_validity.c | 1 - fs/ext4/ext4.h | 2 + fs/ext4/ext4_extents.h | 3 +- fs/ext4/extents.c | 77 +- fs/ext4/fsync.c | 16 +- fs/ext4/inode.c | 225 ++- fs/ext4/mballoc.h | 1 - fs/ext4/super.c | 7 +- fs/ext4/xattr.c | 2 + fs/fcntl.c | 102 +- fs/fs-writeback.c | 18 + fs/gfs2/file.c | 38 +- fs/gfs2/meta_io.c | 2 +- fs/gfs2/ops_inode.c | 6 +- fs/gfs2/xattr.c | 21 +- fs/hppfs/hppfs.c | 18 +- fs/jbd2/checkpoint.c | 15 + fs/jbd2/commit.c | 19 +- fs/jbd2/journal.c | 2 +- fs/namei.c | 24 +- fs/namespace.c | 14 +- fs/nfs/dir.c | 1 + fs/nfsd/vfs.c | 5 +- fs/nilfs2/bmap.c | 4 +- fs/nilfs2/cpfile.c | 31 +- fs/nilfs2/direct.c | 17 +- fs/nilfs2/ioctl.c | 2 +- fs/notify/inotify/inotify_fsnotify.c | 2 +- fs/notify/inotify/inotify_user.c | 4 +- fs/ocfs2/file.c | 21 +- fs/proc/array.c | 89 - fs/proc/base.c | 1 - fs/proc/task_mmu.c | 3 +- fs/quota/dquot.c | 3 + fs/ramfs/file-nommu.c | 26 +- fs/reiserfs/bitmap.c | 3 + fs/reiserfs/inode.c | 24 +- fs/reiserfs/ioctl.c | 3 +- fs/reiserfs/journal.c | 18 +- fs/reiserfs/lock.c | 9 + fs/reiserfs/namei.c | 7 +- fs/reiserfs/xattr.c | 38 +- fs/reiserfs/xattr_acl.c | 2 + fs/sysfs/dir.c | 14 +- fs/sysfs/sysfs.h | 15 + fs/ubifs/gc.c | 96 +- fs/xfs/linux-2.6/xfs_acl.c | 3 +- fs/xfs/linux-2.6/xfs_super.c | 14 +- fs/xfs/linux-2.6/xfs_sync.c | 183 +- fs/xfs/linux-2.6/xfs_sync.h | 2 +- fs/xfs/linux-2.6/xfs_trace.h | 1145 +++++++------ fs/xfs/quota/xfs_qm_syscalls.c | 2 +- fs/xfs/xfs_alloc.c | 44 +- fs/xfs/xfs_dfrag.c | 106 +- fs/xfs/xfs_iget.c | 1 - fs/xfs/xfs_inode.c | 17 +- fs/xfs/xfs_rtalloc.c | 2 + fs/xfs/xfs_vnodeops.c | 93 +- include/acpi/platform/aclinux.h | 2 +- include/drm/drmP.h | 2 +- include/drm/drm_mode.h | 1 + include/drm/i915_drm.h | 54 + include/linux/acpi.h | 7 +- include/linux/blkdev.h | 26 +- include/linux/decompress/unlzo.h | 10 + include/linux/drbd.h | 2 +- include/linux/drbd_nl.h | 1 + include/linux/firewire-cdev.h | 3 + include/linux/firewire.h | 4 +- include/linux/fs.h | 2 +- include/linux/genhd.h | 6 +- include/linux/highmem.h | 2 +- include/linux/i2c/adp5588.h | 12 + include/linux/ieee80211.h | 2 +- include/linux/inetdevice.h | 1 + include/linux/iocontext.h | 27 - include/linux/jbd2.h | 1 + include/linux/kernel.h | 4 + include/linux/kfifo.h | 44 +- include/linux/kgdb.h | 7 +- include/linux/kmemcheck.h | 110 +- include/linux/libata.h | 3 + include/linux/list_sort.h | 11 + include/linux/mfd/tmio.h | 39 + include/linux/mfd/wm8350/pmic.h | 28 +- include/linux/mm.h | 1 + include/linux/mm_types.h | 6 +- include/linux/pci.h | 4 + include/linux/perf_event.h | 11 +- include/linux/phy.h | 1 + include/linux/poison.h | 16 +- include/linux/reiserfs_fs.h | 26 + include/linux/resource.h | 8 +- include/linux/sched.h | 29 +- include/linux/serio.h | 19 - include/linux/string.h | 5 +- include/linux/syscalls.h | 2 +- include/linux/sysctl.h | 1 + include/linux/topology.h | 2 +- include/linux/tty.h | 4 +- include/linux/uaccess.h | 4 +- include/linux/usb.h | 1 + include/linux/writeback.h | 1 + include/net/ip.h | 16 + include/net/mac80211.h | 8 +- include/scsi/libsrp.h | 1 + include/sound/soc-dapm.h | 32 + include/sound/soc.h | 17 + include/trace/ftrace.h | 7 +- init/Kconfig | 18 +- ipc/shm.c | 3 + kernel/cgroup.c | 1 - kernel/futex.c | 27 +- kernel/hw_breakpoint.c | 10 +- kernel/kfifo.c | 107 +- kernel/kmod.c | 12 +- kernel/kprobes.c | 2 +- kernel/module.c | 17 +- kernel/perf_event.c | 14 +- kernel/sched.c | 5 +- kernel/sched_fair.c | 2 +- kernel/signal.c | 3 +- kernel/smp.c | 2 +- kernel/timer.c | 3 +- kernel/trace/Kconfig | 112 +- kernel/trace/ftrace.c | 6 +- kernel/trace/ring_buffer.c | 4 +- kernel/trace/trace.c | 2 +- kernel/trace/trace_events_filter.c | 29 +- kernel/trace/trace_export.c | 7 +- kernel/trace/trace_kprobe.c | 7 +- kernel/trace/trace_ksym.c | 140 +- lib/Kconfig | 4 + lib/Makefile | 3 +- lib/decompress.c | 5 + lib/decompress_unlzo.c | 209 +++ lib/dma-debug.c | 15 +- lib/list_sort.c | 102 ++ lib/lzo/lzo1x_decompress.c | 9 +- lib/rational.c | 1 + lib/string.c | 27 +- lib/vsprintf.c | 4 +- lib/zlib_inflate/inffast.c | 75 +- mm/hugetlb.c | 2 +- mm/maccess.c | 11 +- mm/memcontrol.c | 11 +- mm/mmap.c | 40 + mm/nommu.c | 144 ++- mm/page_alloc.c | 4 +- mm/percpu.c | 4 +- mm/slab.c | 4 +- mm/truncate.c | 30 +- mm/util.c | 46 +- mm/vmalloc.c | 4 +- mm/vmscan.c | 3 + net/bridge/netfilter/ebtables.c | 6 + net/core/pktgen.c | 9 +- net/core/sock.c | 4 + net/ipv4/devinet.c | 1 + net/ipv4/fib_frontend.c | 2 + net/ipv4/ip_output.c | 2 +- net/ipv6/exthdrs.c | 7 +- net/ipv6/ip6_output.c | 3 +- net/mac80211/ht.c | 25 +- net/mac80211/ibss.c | 4 + net/mac80211/iface.c | 47 +- net/mac80211/main.c | 2 + net/mac80211/mlme.c | 16 +- net/mac80211/rx.c | 8 +- net/mac80211/scan.c | 10 +- net/mac80211/tx.c | 9 +- net/mac80211/util.c | 24 + net/mac80211/wme.c | 96 +- net/mac80211/wme.h | 8 +- net/netfilter/ipvs/Kconfig | 3 +- net/netfilter/ipvs/ip_vs_ctl.c | 14 +- net/netfilter/ipvs/ip_vs_wrr.c | 15 +- net/netfilter/nf_conntrack_ftp.c | 18 +- net/packet/af_packet.c | 19 +- net/rose/rose_loopback.c | 2 +- net/sctp/socket.c | 3 +- net/sunrpc/auth_gss/auth_gss.c | 17 +- net/sunrpc/auth_gss/gss_krb5_mech.c | 4 +- net/sunrpc/auth_gss/gss_mech_switch.c | 2 +- net/sunrpc/svc_xprt.c | 3 +- net/wireless/core.c | 4 +- net/wireless/mlme.c | 13 +- net/wireless/reg.c | 2 +- net/wireless/scan.c | 13 +- net/xfrm/xfrm_policy.c | 2 +- scripts/Makefile.lib | 14 +- scripts/checkpatch.pl | 2 +- scripts/get_maintainer.pl | 86 +- scripts/markup_oops.pl | 2 +- scripts/mod/file2alias.c | 2 +- scripts/recordmcount.pl | 2 +- security/selinux/hooks.c | 2 +- security/tomoyo/tomoyo.c | 7 +- sound/arm/aaci.c | 3 +- sound/core/pcm_native.c | 8 +- sound/core/sound.c | 4 +- sound/core/sound_oss.c | 2 +- sound/isa/sb/emu8000.c | 11 +- sound/oss/dev_table.c | 16 +- sound/oss/sound_config.h | 2 + sound/oss/soundcard.c | 4 +- sound/pci/ac97/ac97_codec.c | 10 + sound/pci/ac97/ac97_id.h | 2 + sound/pci/ac97/ac97_patch.c | 1 + sound/pci/atiixp.c | 1 + sound/pci/hda/hda_beep.c | 16 +- sound/pci/hda/hda_codec.c | 10 +- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 21 +- sound/pci/hda/patch_analog.c | 16 + sound/pci/hda/patch_realtek.c | 96 +- sound/pci/hda/patch_sigmatel.c | 79 +- sound/pci/riptide/riptide.c | 2 +- sound/soc/codecs/ad1836.c | 32 + sound/soc/codecs/ad1836.h | 1 + sound/soc/codecs/wm8903.c | 3 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/Makefile | 2 + sound/soc/sh/siu.h | 193 ++ sound/soc/sh/siu_dai.c | 847 +++++++++ sound/soc/sh/siu_pcm.c | 616 +++++++ sound/sound_core.c | 2 +- sound/usb/usbaudio.c | 2 +- tools/perf/Makefile | 52 +- tools/perf/builtin-buildid-list.c | 2 +- tools/perf/builtin-diff.c | 3 +- tools/perf/builtin-kmem.c | 11 +- tools/perf/builtin-record.c | 14 +- tools/perf/builtin-report.c | 3 +- tools/perf/builtin-timechart.c | 2 +- tools/perf/builtin-trace.c | 4 +- tools/perf/design.txt | 8 +- usr/Kconfig | 26 +- virt/kvm/assigned-dev.c | 6 +- virt/kvm/kvm_main.c | 5 +- 1256 files changed, 32865 insertions(+), 11433 deletions(-) rename Documentation/{DMA-mapping.txt => PCI/PCI-DMA-mapping.txt} (100%) delete mode 100644 Documentation/block/as-iosched.txt create mode 100644 Documentation/hwmon/amc6821 copy arch/arm/boot/compressed/{piggy.S => piggy.gzip.S} (67%) rename arch/arm/boot/compressed/{piggy.S => piggy.lzo.S} (68%) create mode 100644 arch/arm/mach-s3c2410/include/mach/ts.h create mode 100644 arch/blackfin/mm/maccess.c delete mode 100644 arch/mips/powertv/cmdline.c create mode 100644 arch/sh/include/asm/siu.h create mode 100644 drivers/gpio/adp5588-gpio.c create mode 100644 drivers/gpu/drm/radeon/reg_srcs/r420 create mode 100644 drivers/hwmon/amc6821.c create mode 100644 drivers/media/dvb/frontends/mb86a16.c create mode 100644 drivers/media/dvb/frontends/mb86a16.h create mode 100644 drivers/media/dvb/frontends/mb86a16_priv.h create mode 100644 drivers/media/dvb/frontends/tda665x.c create mode 100644 drivers/media/dvb/frontends/tda665x.h create mode 100644 drivers/media/dvb/mantis/Kconfig create mode 100644 drivers/media/dvb/mantis/Makefile create mode 100644 drivers/media/dvb/mantis/hopper_cards.c create mode 100644 drivers/media/dvb/mantis/hopper_vp3028.c create mode 100644 drivers/media/dvb/mantis/hopper_vp3028.h create mode 100644 drivers/media/dvb/mantis/mantis_ca.c create mode 100644 drivers/media/dvb/mantis/mantis_ca.h create mode 100644 drivers/media/dvb/mantis/mantis_cards.c create mode 100644 drivers/media/dvb/mantis/mantis_common.h create mode 100644 drivers/media/dvb/mantis/mantis_core.c create mode 100644 drivers/media/dvb/mantis/mantis_core.h create mode 100644 drivers/media/dvb/mantis/mantis_dma.c create mode 100644 drivers/media/dvb/mantis/mantis_dma.h create mode 100644 drivers/media/dvb/mantis/mantis_dvb.c create mode 100644 drivers/media/dvb/mantis/mantis_dvb.h create mode 100644 drivers/media/dvb/mantis/mantis_evm.c create mode 100644 drivers/media/dvb/mantis/mantis_hif.c create mode 100644 drivers/media/dvb/mantis/mantis_hif.h create mode 100644 drivers/media/dvb/mantis/mantis_i2c.c create mode 100644 drivers/media/dvb/mantis/mantis_i2c.h create mode 100644 drivers/media/dvb/mantis/mantis_input.c create mode 100644 drivers/media/dvb/mantis/mantis_ioc.c create mode 100644 drivers/media/dvb/mantis/mantis_ioc.h create mode 100644 drivers/media/dvb/mantis/mantis_link.h create mode 100644 drivers/media/dvb/mantis/mantis_pci.c create mode 100644 drivers/media/dvb/mantis/mantis_pci.h create mode 100644 drivers/media/dvb/mantis/mantis_pcmcia.c create mode 100644 drivers/media/dvb/mantis/mantis_reg.h create mode 100644 drivers/media/dvb/mantis/mantis_uart.c create mode 100644 drivers/media/dvb/mantis/mantis_uart.h create mode 100644 drivers/media/dvb/mantis/mantis_vp1033.c create mode 100644 drivers/media/dvb/mantis/mantis_vp1033.h create mode 100644 drivers/media/dvb/mantis/mantis_vp1034.c create mode 100644 drivers/media/dvb/mantis/mantis_vp1034.h create mode 100644 drivers/media/dvb/mantis/mantis_vp1041.c create mode 100644 drivers/media/dvb/mantis/mantis_vp1041.h create mode 100644 drivers/media/dvb/mantis/mantis_vp2033.c create mode 100644 drivers/media/dvb/mantis/mantis_vp2033.h create mode 100644 drivers/media/dvb/mantis/mantis_vp2040.c create mode 100644 drivers/media/dvb/mantis/mantis_vp2040.h create mode 100644 drivers/media/dvb/mantis/mantis_vp3028.c create mode 100644 drivers/media/dvb/mantis/mantis_vp3028.h create mode 100644 drivers/media/dvb/mantis/mantis_vp3030.c create mode 100644 drivers/media/dvb/mantis/mantis_vp3030.h create mode 100644 drivers/mfd/tmio_core.c delete mode 100644 drivers/s390/cio/qdio_perf.c delete mode 100644 drivers/s390/cio/qdio_perf.h mode change 100644 => 100755 drivers/scsi/lpfc/lpfc_hw4.h create mode 100644 firmware/cis/PE520.cis.ihex create mode 100644 include/linux/decompress/unlzo.h create mode 100644 include/linux/list_sort.h create mode 100644 lib/decompress_unlzo.c create mode 100644 lib/list_sort.c create mode 100644 sound/soc/sh/siu.h create mode 100644 sound/soc/sh/siu_dai.c create mode 100644 sound/soc/sh/siu_pcm.c

2 1

[alsa-devel] Samsung: New driver for AC97 controllers
by jassisinghbrar＠gmail.com 26 Jan '10

26 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> Hi, I am going to submit new driver for the AC97-ver-2.0 controllers found in Samsung SoCs. The driver is named generic s3c-ac97.c The driver has been tested for on s3c64xx and newer SoCs and is confirmed to support Playback/Capture and VRA. A machine driver, for SMDKs with wm9713 as AC97 codec attached, has also been added. The driver is generic enough to be used as such with newer SoCs. The controller driver patch is meant for Mark Brown's for-2.6.34 Other patches apply cleanly against the Ben Dooks' next-samsung The s3c-ac97.c is meant to replace s3c2443-ac97.c after someone confirms Playback/Capture to work fine. My smdk2443 seems to not play any sound though the DAI link is establised and the codec detected fine. So, my opinion is that the new driver is atleast as good as the old one. If that is good enough, I have patches ready that switch the only two users of s3c2443-ac97.c(smdk2443 and ln2440sbc) to use this new driver. Regards.

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[alsa-devel] [PATCH 5/5] DaVinci: DM365: Voice Codec support for the DM365 EVM
by miguel.aguilar＠ridgerun.com 26 Jan '10

26 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec, the idea is to have both enabled in the same kernel simultaneously. However, the current soc-core doesn't support simultaneous codecs, once that support will have added, a patch will be posted to enable both codecs in the DM365 EVM. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 +++ sound/soc/davinci/Kconfig | 24 +++++++++++++++- sound/soc/davinci/davinci-evm.c | 45 +++++++++++++++++++++++++++++- 3 files changed, 69 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 5c2636c..865d06a 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -516,7 +516,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#ifdef CONFIG_SND_DM365_AIC3X_CODEC dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + dm365_init_vc(&dm365_evm_snd_data); +#endif dm365_init_rtc(); dm365_init_ks(&dm365evm_ks_data); } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 47e7cce..6bbf001 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -18,12 +18,32 @@ config SND_DAVINCI_SOC_VCIF config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..ef63096 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,10 +28,12 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" +#include "davinci-vcif.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -151,6 +153,22 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai = { +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +#endif +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +195,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +203,15 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = 1, +}; + + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +244,17 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +282,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; -- 1.6.0.4

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[alsa-devel] [PATCHv2 1/6] ASoC: OMAP: data_type and sync_mode configurable in audio dma
by Candelaria Villareal, Jorge 25 Jan '10

25 Jan '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 ++ sound/soc/omap/omap-pcm.c | 15 ++++++++------- sound/soc/omap/omap-pcm.h | 4 +++- 3 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2a..d297256 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770..825db38 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821d..b19975d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 25 Jan '10

25 Jan '10

The following changes since commit 92dcffb916d309aa01778bf8963a6932e4014d07: Linus Torvalds (1): Linux 2.6.33-rc5 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The pull request includes a much bigger delta against Linus' tree, I merged up since the last batch of fixes that went into his tree didn't seem to have shown up on your fix branch for some reason. Guennadi Liakhovetski (1): ASoC: fix a memory-leak in wm8903 sound/soc/codecs/wm8903.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH v2] ad1836: reset and restore clock control mode in suspend/resume entry
by Barry Song 25 Jan '10

25 Jan '10

tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- -v2: commit forgotten ad1836.h sound/soc/codecs/ad1836.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1836.h | 1 + 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 2c18e3d..83add2f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -223,6 +223,36 @@ static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, return reg_cache[reg]; } +#ifdef CONFIG_PM +static int ad1836_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* reset clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} + +static int ad1836_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* restore clock control mode */ + u16 adc_ctrl2 = codec->read(codec, AD1836_ADC_CTRL2); + adc_ctrl2 |= AD1836_ADC_AUX; + + return codec->write(codec, AD1836_ADC_CTRL2, adc_ctrl2); +} +#else +#define ad1836_soc_suspend NULL +#define ad1836_soc_resume NULL +#endif + static int __devinit ad1836_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -404,6 +434,8 @@ static int ad1836_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_ad1836 = { .probe = ad1836_probe, .remove = ad1836_remove, + .suspend = ad1836_soc_suspend, + .resume = ad1836_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 7660ee6..e9d90d3 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -54,6 +54,7 @@ #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) +#define AD1836_ADC_AUX (0x6 << 6) #define AD1836_ADC_CTRL3 14 -- 1.5.6.3

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[alsa-devel] [PATCHv2 0/6] McPDM driver patch series
by Candelaria Villareal, Jorge 25 Jan '10

25 Jan '10

McPDM module is the interface between the OMAP and TWL6030 audio codec. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. Changes from v1: - Device registration now takes place plat-omap/devices.c - struct resources is used to store mem and irq resources. - DMA allows packet_size to be configured. - PACKET is now used as sync_mode for McPDM -- Jorge Eduardo Candelaria (4): OMAP4: IRQ: Add McPDM IRQ definition. ARM: OMAP4: Add McPDM base address OMAP4: MCPDM: Register McPDM platform device ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver arch/arm/plat-omap/devices.c | 34 ++ arch/arm/plat-omap/include/plat/irqs.h | 1 + arch/arm/plat-omap/include/plat/omap44xx.h | 2 + sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 524 ++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 153 ++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 248 +++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 ++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- 12 files changed, 1009 insertions(+), 8 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c

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[alsa-devel] snd-hda-intel w/Realtek ALC260 codec broken on FJS Celsius M440
by Martin Lucina 25 Jan '10

25 Jan '10

Hello, in the interests of helping to debug the mess that is snd-hda-intel and getting sound working properly on my machine, I'd like to report the following problem: This machine is a Fujitsu Siemens Celsius M440 desktop -- about 4 years old and running Debian lenny. The sound is onboard Intel HD Audio with a Realtek ALC260 codec. The mainboard (Fujitsu D2178) has the following jacks: Rear: (left to right) Pink - MIC - Node 0x12 in hda-analyzer Green - Speaker/Line out(?) - Node 0x0f in hda-analyzer Blue - Line in - Node 0x14 in hda-analyzer Front: (left to right) Green - Headphone - Node 0x15 in hda-analyzer Pink - MIC - Node 0x13 in hda-analyzer Internal: (at least, can confirm detail if needed) CD Internal speaker The current status with kernel 2.6.32.5 is as follows: 1) snd_hda_intel loaded with model=basic (the "slightly less broken" option) Mixer playback behaviour: Headphone -> nothing Front -> Front and Rear green jack Line -> Not tested CD -> Not tested Beep -> Beep Capture behaviour: I've only tested with a headset connected to the front jacks, with "Capture" set to 100% and "Capture Source" set to "Front Mic". In this mode I seem to get a *very weak* signal from the mic as tested by doing "arecord > ./test.wav", then "aplay ./tmp.wav". 2) snd_hda_intel loaded with model=auto (the "completely broken" option) Mixer playback behaviour: Headphone -> Internal speaker Front -> Rear green jack Beep -> Beep Capture: untested In this mode the front jacks don't seem to work at all. I'd appreciate any help on debugging this. The ideal configuration which I guess should(?) be possible is separate mixers for Front out, Rear out, Internal speaker, and working capture with both Front and Rear Mic. Also, can anyone suggest a better tool to test recording than "arecord"? Specifically something that will actually tell me if the level of the sound recorded from the Mic is "sane"? Also see the attached output from alsa-info.sh for both model=auto and model=basic. I tried a few other models but with no luck. Thanks, -mato

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[alsa-devel] [PATCH 0/5] Add Voice Codec support for DaVinci platforms
by Miguel Aguilar 25 Jan '10

25 Jan '10

Hi, The following patch series adds support for the Voice Codec in the Davinci platform and in this particular case for the DM365 EVM. The DM365 EVM has two codecs the AIC3x and the Voice Codec, however at this point is not possible to have both enabled simultaneously since the soc-core doesn't support that, so a patch for enable both simultaneously will be sent when this support is available in the soc-core. Regards, -- Miguel Angel Aguilar Ulloa Embedded Software Engineer RidgeRun Embedded Solutions miguel.aguilar(a)ridgerun.com

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[alsa-devel] [PATCH 4/5] DaVinci: DM365: Voice codec support for the DM365 SoC
by miguel.aguilar＠ridgerun.com 25 Jan '10

25 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the generic Voice Codec support for the DM365 based platforms. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 37 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/dm365.h | 5 ++++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 43 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index c63afc0..b5345d5 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct clk_lookup dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -606,6 +606,8 @@ INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) #endif }; @@ -835,6 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -991,6 +1018,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { dm365_ks_device.dev.platform_data = pdata; diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..57e51eb 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 137bfba..2a68c1d 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -327,6 +327,8 @@ enum davinci_dm365_index { /* EDMA event muxing */ DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT26_MMC0_RX, }; -- 1.6.0.4

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[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 25 Jan '10

25 Jan '10

Dear ALSA Team, I have some queries.Please answer them.(previous post please ignore as they couldn't appear because of encoding problem) I am getting underrun problem whenever the playback is about to get finished. happening in snd_pcm_update_hw_ptr_post function: if (avail >= stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); //here it is happening.......... return -EPIPE; } 1. As i understand if i dont call drain function this will happen.Ismy understanding correct?Because if am calling drain function then thisproblem is not happening.If the application doesn't call drain functionthen xrun is the only way to stop ALSA or does ALSA detect that thereare no more samples to play and automatically stop the play(if yes,canyou point out the source code)? 2. snd_pcm_playback_avail() returns the number of samples that stillneeds to be played?Below is the diagram which is my understanding. Ring buffer (DMA buffer created using preallocated buffer during system initalisation) ----------------------------------------------------------------- | | | | | | | ----------------------------------------------------------------- ^ ^ | | hw_ptr appl_ptr After first write call: ----------------------------------------------------------------- | | | | | | | ----------------------------------------------------------------- ^ ^ | | appl_ptr hw_ptr So my understanding is once wait_for_avail_min returns insnd_pcm_lib_write1 function.hw_ptr is increased by some amount(periodsize) but appl_ptr will increase only when wait_for_avail_minreturns.And difference of (hw_ptr+runtime->buffersize-appl_ptr) will give us unplayed samples in the case of playback? Is this whole ring buffer is filled after the snd_pcm_lib_write_transfer function is returned? 3.In some rare cases (playback case) i have seen thatwait_for_avail_min doesn't return and driver pointer function keeps ongetting called causing hw_ptr to increase.This causessnd_pcm_playback_avail() function to return a value less than avail_min(as hw_ptr is increasing and appl_ptr is stagnant as it is stuck insnd_pcm_lib_write1 function) which inturn causes wake-up fromsnd_pcm_update_hw_ptr_post() not to get called.When this type ofscenario happens?Is it a normal scenario if yes,then how will it willever come out from wait_for_avail_min? According to my understanding appl_ptr will increase only when we return from wait_for_avail_min? 4.how does alsa know that it is running short of data?As iunderstand that it does not free the memory and data is onlyoverwritten.So how does it distinguish between old and new data? 5.Burst of noise due to stale data problem reported by Jon Smirl and for which Mr Jaroslav has suggested this solution http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/017119.html.Is it implemented? Best Regards..

2 1

[alsa-devel] Underrun problem and some basic queries
by ANISH KUMAR 25 Jan '10

25 Jan '10

Hi Takashi et al, This post of mine is just a reminder about my earlier unanswered questions which i raised about underrun. http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024286.ht… I guess you were on holiday and i think this is a bit long list of questions which no body cared to answer.This time please answer them as this would help all the new entrants to understand ALSA. Best Regards, anish

1 0

[alsa-devel] How much of the X-Fi (emu20k1) chip has Creative documented?
by Chris Rankin 25 Jan '10

25 Jan '10

Hi, I am thinking about upgrading my Audigy2 sound card, and was looking at the X-Fi. However, the feature matrix for X-Fi cards on the ALSA site only says "PCI". (I assume that also includes PCIe these days.) Given that there is a basic snd-ctxfi driver, it seems likely that this matrix should also include "ANALOGio", and so I was wondering if someone could tell me what the actual matrix entry for the X-Fi cards should be these days, please? Would the documentation that Creative has released allow any additional features to be implemented in the future? I am obviously hoping at least for feature parity with the Audigy2, i.e. multi-voiced hardware mixing. That "crystalizer" would seem to be one of the cards main selling points, too. However, if the driver is "feature complete" as far as Creative's documentation goes, then one of the Xonar cards would seem better value for money. Thanks for any assistance, Cheers, Chris

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Re: [alsa-devel] Reg: Audio recording with twl4030 codec
by Peter Ujfalusi 25 Jan '10

25 Jan '10

Hello, Let's put alsa-devel back to the CC... On Monday 25 January 2010 09:32:45 ext shariff wrote: > Hi Peter, > > 1) As we only have one Main MIC when I am recording in stereo mode by > specifying "-c 2", one channel is silent and one channel is recorded > properly. As it should. Just as a sidenote, I'm usually using -f dat for the format, which gives 48KHz, stereo 16LE > > 2) But still the recorded channel is noisy, how can I reduce the noise of > the recorded channel? Hmm, what kind of noise? How are you configuring the dai with the snd_soc_dai_set_fmt? For stereo it should be (if OMAP is slave, TWL is master): SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; For TDM mode (four channel): SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; See omap3beagle.c how it is done. > > I have one more question. ( Regarding Audio path) > > ---> If I want to avoid that one channel silent, do I need to connect "SUB > MIC or AUXR"?( as these are connected right ADC). If you want audio data there, than yes you need another mic for the right channel. > ----> If I need to record 4 channel audio without any silent channel do I > need to connect digital microphones also? You can duplicate the channel 1 and 3 to channel 2 and 4, meaning that you route the ADCL and ADCR to these channels (TX1 and TX2 path are both analog). You can also hook up a stereo digital mic to the digimic1 interface, but remember, that the digimic needs 48KHz sampling rate. -- Péter > > Regards, > Shariff > > On Fri, Jan 22, 2010 at 6:29 PM, shariff > <shariffsinbox(a)gmail.com<mailto:shariffsinbox@gmail.com>> wrote: Hi Peter, > > Thank you very much for your suggestions. > > Regards, > Shariff > > > On Fri, Jan 22, 2010 at 6:11 PM, Peter Ujfalusi > <peter.ujfalusi(a)nokia.com<mailto:peter.ujfalusi@nokia.com>> wrote: Hello, > > On Friday 22 January 2010 11:57:23 ext shariff wrote: > > Hi all, > > > > I am working with OMAP3 based custom board that has twl4030 codec > > & using linux-2.6.29 kernel from opensource. > > This is kind of old... > > > I have some queries regarding Audio recording. > > > > Audio codec is connected to twl4030 as bellow > > > > TWL4030 > > > > |--------------------| > > | > > | |------ MAINMIC > > > > McBSP2 ------ |i2s/tdm | > > > > | ___________| > > > > --> One Analog MIC is connected to Main Mic and I am using Audio > > path not the Voice path. > > > > After selecting Main Mic through amixer control. > > > > and recording audio using "arecord -c 1 -r 8000 -f S16_LE -t > > wav -d 20 record-file.wav" > > The hardware should have been opened in stereo mode, since it is not > supporting the mono, you can verify it by appending -v to arecord and see > how the HW has been opened at the end. > > > --> Here record is successful but with lot of noise. > > > > 1) Can we record Mono(1 channel) through I2S from TWL4030 > > codec? > > The TWL codec "Audio" path is either stereo or 4 channel. > You need to record stereo/4 channel, and than extract/use the data from the > valid channel. > In case of Main mic it is the left channel, or Channel 1 or/and 2 > (depending on the routings) in four channel mode. > > > 2) To record stereo(2 channels), do we need two Mic's ? (i.e do > > I need Sub Mic also). > > No you don't need. The right channel will not contain valid data (most > likely it is going to be silent). > > You need to enable the Mic bias 1 for the main mic in your machine driver, > if you have not done that yet: > > static const struct snd_soc_dapm_widget your_widgets[] = { > ... > SND_SOC_DAPM_MIC("Your Mic", NULL), > ... > }; > > static const struct snd_soc_dapm_route your_audio_map[] = { > ... > {"MAINMIC", NULL, "Mic Bias 1"}, > {"Mic Bias 1", NULL, "Your Mic"}, > ... > }; > > I'm not sure about the state of the twl soc codec driver in 2.6.29, but my > guess is that it should work. > > Are there any capture routing option for TX1 path, which selects between > Analog and Digimic0? If there is than that has to be set to analog as > well. > > > Regards, > > Shariff > > -- > Péter >

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 25 Jan '10

25 Jan '10

The following changes since your topic/asoc branch are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Since they include (via the 2.6.33 merge) a merge of Linus' tree I've only included the changes that aren't already in your tree or mainline in the summary below, though it looks like the diffstat picked up some extra changes due to the way I tried to do that. Barry Song (4): ASoC: ad1836: reset and restore clock control mode in suspend/resume entry ASoC: ad1938: fix typo, rename mask to rx_mask for ad1938_set_tdm_slot ASoC: ad1938: let soc-core dapm handle PLL power ASoC: soc-cache: cleanup training whitespace and coding style Ilkka Koskinen (2): ASoc: tpa6130a2: Remove unnecessary variable ASoC: tpa6130a2: Support for tpa6140's regulators Kuninori Morimoto (3): ASoC: fsi: Add over_period flag to prevent the misunderstanding ASoC: fsi: Add fsi_get_dai to get snd_soc_dai ASoC: fsi: Add over/under run error settlement Mark Brown (9): ASoC: Remove unneeded suspend checks from CODEC drivers ASoC: Use snprintf() when generating stream names ASoC: Fix variable shadowing warning in TLV320AIC3x ASoC: Only restore non-default registers for WM8731 ASoC: Only restore non-default registers for WM8776 ASoC: Only restore non-default registers for WM8961 ASoC: Implement suspend and resume for WM8993 ASoC: Fix WM8350 DSP mode B configuration Merge branch 'for-2.6.33' into for-2.6.34 Peter Ujfalusi (5): ASoC: tlv320dac33: Change nsample switch to FIFO mode enum ASoC: tlv320dac33: Introduce prefill and playback state handlers ASoC: tlv320dac33: Clean up the hardware configuration code ASoC: tlv320dac33: Add new FIFO mode: mode 7 ASoC: tlv320dac33: Safety check for codec slave mode include/sound/soc.h | 3 + include/sound/tpa6130a2-plat.h | 6 + include/sound/wm8904.h | 57 + include/sound/wm8955.h | 26 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 10 +- sound/soc/codecs/ad1836.c | 32 + sound/soc/codecs/ad1938.c | 64 +- sound/soc/codecs/cs4270.c | 43 +- sound/soc/codecs/da7210.c | 589 +++++++++ sound/soc/codecs/da7210.h | 24 + sound/soc/codecs/tlv320aic3x.c | 75 +- sound/soc/codecs/tlv320dac33.c | 282 ++++- sound/soc/codecs/tpa6130a2.c | 109 ++- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8727.c | 66 +- sound/soc/codecs/wm8731.c | 3 + sound/soc/codecs/wm8753.c | 8 - sound/soc/codecs/wm8776.c | 2 + sound/soc/codecs/wm8904.c | 2556 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8904.h | 1681 ++++++++++++++++++++++++ sound/soc/codecs/wm8955.c | 1151 +++++++++++++++++ sound/soc/codecs/wm8955.h | 489 +++++++ sound/soc/codecs/wm8961.c | 3 + sound/soc/codecs/wm8990.c | 8 - sound/soc/codecs/wm8993.c | 67 + sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-mcasp.h | 1 + sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 120 +-- sound/soc/sh/Kconfig | 8 + sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 ++ sound/soc/sh/fsi.c | 164 ++- sound/soc/soc-cache.c | 2 +- sound/soc/soc-core.c | 11 +- sound/soc/soc-dapm.c | 37 +- 37 files changed, 7457 insertions(+), 359 deletions(-) create mode 100644 include/sound/wm8904.h create mode 100644 include/sound/wm8955.h create mode 100644 sound/soc/codecs/da7210.c create mode 100644 sound/soc/codecs/da7210.h create mode 100644 sound/soc/codecs/wm8904.c create mode 100644 sound/soc/codecs/wm8904.h create mode 100644 sound/soc/codecs/wm8955.c create mode 100644 sound/soc/codecs/wm8955.h create mode 100644 sound/soc/sh/fsi-da7210.c

4 19

[alsa-devel] ALSA: hda - add possibility to choose speakers configuration for 4930g
by Łukasz Wojniłowicz 25 Jan '10

25 Jan '10

I've got such possibility on MS Windows XP with Realtek drivers (prove is at http://img694.imageshack.us/i/choosechannelmod.png/ ) so I should have possibility to do it also on Linux e.g. here http://img704.imageshack.us/i/screenshotsoundpreferen.png/ Patch given in attachment should do the thing. patch_realtek.c was downloaded from http://git.alsa-project.org/?p=alsa-kernel.git on Thu, 21 Jan 2010 13:58:25 +0000

2 1

Re: [alsa-devel] snd_pcm_hw_params_free() immediately after snd_pcm_hw_params() ???
by Clemens Ladisch 24 Jan '10

24 Jan '10

Raymond Yau wrote: > Is there any flags in substream which can indicate snd_xxx_hw_params() was > called at first time since snd_xxx_hw_params() were called many times > during OSS emulation (probing the supported rates , channels , .... ) ? There is no such flag. It is the responsibility of the driver to track the resources it has allocated. If a resource does not depend on the hardware parameters, it might be a better idea to allocate/free it in the open/close callbacks. Best regards, Clemens

3 2

[alsa-devel] [PATCH] add simplified versions of widget macros
by Guennadi Liakhovetski 23 Jan '10

23 Jan '10

Many macros from include/sound/soc-dapm.h take an array and a number of elements in it as arguments, whereas most users use static arrays and use "x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of those macros, calling ARRAY_SIZE() internally. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5c95e1..c0922a0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -95,6 +95,21 @@ .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ .num_kcontrols = 1} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\ + wcontrols) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)} +#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols)\ +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols)} + /* path domain with event - event handler must return 0 for success */ #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ wncontrols, wevent, wflags) \ @@ -126,6 +141,23 @@ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ .event = wevent, .event_flags = wflags} +/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ +#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \ + .event = wevent, .event_flags = wflags} +#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \ + wcontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags} + /* events that are pre and post DAPM */ #define SND_SOC_DAPM_PRE(wname, wevent) \ { .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \

3 2

[alsa-devel] [PATCH] ASoC: fix a memory-leak in wm8903
by Guennadi Liakhovetski 23 Jan '10

23 Jan '10

Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- should go in -rc, maybe also for stable diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce5515e..3595bd5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev) struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults), + u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), GFP_KERNEL); /* Bring the codec back up to standby first to minimise pop/clicks */ @@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev) for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) snd_soc_write(codec, i, tmp_cache[i]); + kfree(tmp_cache); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); }

3 2

[alsa-devel] ca0106 broken (fwd)
by Russell Whitaker 23 Jan '10

23 Jan '10

Hi Sent msg below and no responce. As a work-around replaced ng code with know good code into 2.6.29.6 and it worked. Tried same trick with 2.6.31.12 and it won't compile. If you send me something to try will do it and report back. Russ ---------- Forwarded message ---------- Date: Wed, 21 Oct 2009 14:54:45 -0700 (PDT) From: Russell Whitaker <russ(a)ashlandhome.net> To: James(a)superbug.demon.co.uk Subject: ca0106 broken Hi ca0106 is known to work in these releases: linux-2.6.27.20 linux-2.6.27.23 linux-2.6.28.10 It is broken in: linux-2.6.29.6 linux-2.6.30.9 For the broken releases dmesg reports: unknown hardware "CA0106" "" "" "" "" Anything I can do to help? Russ

2 3

Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 23 Jan '10

23 Jan '10

Hello Giuliano, No luck, same behavior. I edited the first file listed listed below and recompiled it and installed it. Then I restarted the system with the indigo card in place, and the VU meters show levels in the -120 to -100 dB? I do have alsa-driver-1.0.22 my kernel is 2.6.31 // Salvador ________________________________ From: Giuliano Pochini <pochini(a)shiny.it> To: S. Aguinaga <sa_jr(a)yahoo.com> Sent: Sun, January 17, 2010 4:04:38 PM Subject: Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo On Thu, 14 Jan 2010 12:13:55 -0800 (PST) "S. Aguinaga" <sa_jr(a)yahoo.com> wrote: > Hello Fellows, > Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. First install a recent version of echomixer. If you are using alsa-driver 1.0.22 or kernel 2.6.32, try to add this line: #define ECHOCARD_HAS_LINE_OUT_GAIN in alsa-driver-1.0.22/alsa-kernel/pci/echoaudio/indigoio.c or in linux-2.6.32/sound/pci/echoaudio/indigoio.c and recompile. It re-enables the line-out control that I removed last year because it was not supposed to exist in that card. -- Giuliano.

2 2

[alsa-devel] [PATCH] modem.conf Off-hook improve behavior
by David Fries 23 Jan '10

23 Jan '10

From: David Fries <david(a)fries.net> Only restore the old value if it differs from the requested value, because if it has changed restoring the old value overrides the change. Take for example, a voice modem with a .conf that sets preserve off-hook. Start playback (on-hook to off-hook), start record (off-hook to off-hook), stop playback (off-hook to restore on-hook), stop record (on-hook to restore off-hook), Clearly you don't want to leave the modem "on the phone" now that there isn't any playback or recording active. Signed-off-by: David Fries <david(a)fries.net> --- Comments welcome. The default modem.conf has, hooks.0 { type ctl_elems hook_args [ { name "Off-hook Switch" preserve true value "on" lock false optional true } ] } I wrote the patch to be, if you didn't modify it, don't restore the value. I also thought that it might work to check to see if the current mixer value matches the requested value (which is the value the program set on open), and only set the old value if it hasn't been externally modified. The theory being if something else changed the mixer value, we don't want to by default overwrite that change. Some additional flags could also be added to leave the current behavior alone and add the if you didn't modify it, don't restore it, or if it changed after you set it, don't restore it. Here are some flag names if that would be preferred, preserve_if_different restore_if_unmodified no_squash restore_overrides preserve_if_changed. Setting the lock isn't really an option, because then you can't manually "hang up" the modem if you needed to. On a side note, do any of the alsa voice/software modems support notifying userspace when the line is ringing? I'm using intel8x0m, not that I think it matters here. diff --git a/include/control.h b/include/control.h index 2361dc3..ded884a 100644 --- a/include/control.h +++ b/include/control.h @@ -419,6 +419,7 @@ int snd_ctl_elem_value_malloc(snd_ctl_elem_value_t **ptr); void snd_ctl_elem_value_free(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_clear(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value_t *src); +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right); void snd_ctl_elem_value_get_id(const snd_ctl_elem_value_t *obj, snd_ctl_elem_id_t *ptr); unsigned int snd_ctl_elem_value_get_numid(const snd_ctl_elem_value_t *obj); snd_ctl_elem_iface_t snd_ctl_elem_value_get_interface(const snd_ctl_elem_value_t *obj); diff --git a/src/control/control.c b/src/control/control.c index df249dd..159b09a 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2256,6 +2256,18 @@ void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value } /** + * \brief compare one #snd_ctl_elem_value_t to another + * \param dst pointer to destination + * \param src pointer to source + * \return 0 on match, less than or greater than otherwise, see memcmp + */ +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right) +{ + assert(left && right); + return memcmp(left, right, sizeof(*left)); +} + +/** * \brief Get CTL element identifier of a CTL element id/value * \param obj CTL element id/value * \param ptr Pointer to returned CTL element identifier diff --git a/src/control/setup.c b/src/control/setup.c index 408244e..3606daf 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -192,7 +192,17 @@ int snd_sctl_remove(snd_sctl_t *h) return err; } } - if (elem->preserve) { + /* Only restore the old value if it differs from the requested + * value, because if it has changed restoring the old value + * overrides the change. Take for example, a voice modem with + * a .conf that sets preserve off-hook. Start playback (on-hook + * to off-hook), start record (off-hook to off-hook), stop + * playback (off-hook to restore on-hook), stop record (on-hook + * to restore off-hook), Clearly you don't want to leave the + * modem "on the phone" now that there isn't any playback or + * recording active. + */ + if (elem->preserve && snd_ctl_elem_value_compare(elem->val, elem->old)) { err = snd_ctl_elem_write(h->ctl, elem->old); if (err < 0) { SNDERR("Cannot restore ctl elem");

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[alsa-devel] default alsa pcm improvement
by Aleksandar Kostadinov 23 Jan '10

23 Jan '10

Hello, _From some time now ALSA is a real pleasure to use. Namely it works by OOB and has lots of predefined useful PCMs so one doesn't need to be a guru to do advanced operations. I think that a small improvement in how default PCM is defined can help users with multiple sound cards that like to use pure ALSA and not some of the many available sound servers. I want to be able to switch the default sound device easily in asoundrc but the way the default pcm is defined makes this not very straightforward. I like to use the definitions of default:<CARD> but if I define "pcm.!default" in .asoundrc I lose the "default:<CARD>" definition. I now can rename the "default" definition in default.conf to someting like "defaultorg" and then reference it asoundrc but that requires root as well change can possibly disappear after an alsa-lib package upgrade. Or I can use the dmix and dsnoop predefined PCMs to put together PCMs that I can later reference in my "pcm.!default" definition but in this case one needs to know if the card needs dsnoop/dmix and probably other tweaks that default.conf contains. Additionally this needs to be done for every sound card one owns. So I would suggest the "default" PCM as defined now in the default.conf file to be renamed to something like "alsadefault". Then "default" PCM should be an alias to the alsadefault pcm. This will be backwards compatible to what users have now and owners of multiple sound devices can define their "default" pcm in asoundrc referencing the alsadefault pcm benefiting from the good work already done and not have to mess with a complex asoundrc file or make changes to the system configurations files. Please CC me in replies because I'm not subscribed. Thanks again for the cool work!

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[alsa-devel] [PATCHv2 4/6] OMAP4: MCPDM: Register McPDM platform device
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Add platform device for McPDM interface in OMAP4 and register its memory and irq resources. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/devices.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/devices.c b/arch/arm/plat-omap/devices.c index 30b5db7..572265c 100644 --- a/arch/arm/plat-omap/devices.c +++ b/arch/arm/plat-omap/devices.c @@ -28,6 +28,7 @@ #include <plat/menelaus.h> #include <plat/mcbsp.h> #include <plat/dsp_common.h> +#include <plat/omap44xx.h> #if defined(CONFIG_OMAP_DSP) || defined(CONFIG_OMAP_DSP_MODULE) @@ -192,6 +193,41 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, /*-------------------------------------------------------------------------*/ +#if defined(CONFIG_SND_OMAP_SOC_MCPDM) || \ + defined(CONFIG_SND_OMAP_SOC_MCPDM_MODULE) + +static struct resource mcpdm_resources[] = { + { + .name = "mcpdm_mem", + .start = OMAP44XX_MCPDM_BASE, + .end = OMAP44XX_MCPDM_BASE + SZ_4K, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcpdm_irq", + .start = INT_44XX_MCPDM_IRQ, + .end = INT_44XX_MCPDM_IRQ, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device omap_mcpdm_device = { + .name = "omap-mcpdm", + .id = -1, + .num_resources = ARRAY_SIZE(mcpdm_resources), + .resource = mcpdm_resources, +}; + +static void omap_init_mcpdm(void) +{ + (void) platform_device_register(&omap_mcpdm_device); +} +#else +static inline void omap_init_mcpdm(void) {} +#endif + +/*-------------------------------------------------------------------------*/ + #if defined(CONFIG_MMC_OMAP) || defined(CONFIG_MMC_OMAP_MODULE) || \ defined(CONFIG_MMC_OMAP_HS) || defined(CONFIG_MMC_OMAP_HS_MODULE) @@ -385,6 +421,7 @@ static int __init omap_init_devices(void) omap_init_dsp(); omap_init_kp(); omap_init_rng(); + omap_init_mcpdm(); omap_init_uwire(); omap_init_wdt(); return 0; -- 1.6.0.4

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[alsa-devel] [PATCHv2 3/6] ARM: OMAP4: Add McPDM base address
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Define McPDM physical and L3 base address for OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/omap44xx.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/omap44xx.h b/arch/arm/plat-omap/include/plat/omap44xx.h index 8e4c581..4c64f20 100644 --- a/arch/arm/plat-omap/include/plat/omap44xx.h +++ b/arch/arm/plat-omap/include/plat/omap44xx.h @@ -43,6 +43,8 @@ #define OMAP44XX_SCU_BASE 0x48240000 #define OMAP44XX_LOCAL_TWD_BASE 0x48240600 #define OMAP44XX_WKUPGEN_BASE 0x48281000 +#define OMAP44XX_MCPDM_BASE 0x40132000 +#define OMAP44XX_MCPDM_L3_BASE 0x49032000 #define OMAP44XX_MAILBOX_BASE (L4_44XX_BASE + 0xF4000) -- 1.6.0.4

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[alsa-devel] [PATCHv2 2/6] OMAP4: IRQ: Add McPDM IRQ definition.
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Support for McPDM IRQs in OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/irqs.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index c0ab7c8..bbb8eb6 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -432,7 +432,7 @@ #define INT_44XX_PARTHASH_IRQ (79 + IRQ_GIC_START) #define INT_44XX_MMC3_IRQ (94 + IRQ_GIC_START) #define INT_44XX_MMC4_IRQ (96 + IRQ_GIC_START) - +#define INT_44XX_MCPDM_IRQ (112 + IRQ_GIC_START) /* Max. 128 level 2 IRQs (OMAP1610), 192 GPIOs (OMAP730/850) and * 16 MPUIO lines */ -- 1.6.0.4

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[alsa-devel] Conexant 5051 subsystem 0x103c30d6
by Jacopo De Simoi 22 Jan '10

22 Jan '10

Dear all, this is a follow up to my previous posting to the ml about having CX 5051 working on my laptop; I found out that the patch provided in http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024284.ht… fits perfectly my model as well. Hence I suggest to add a quirk matching my subsystem id as follows: SND_PCI_QUIRK(0x103c, 0x30d6, "HP DV2700", CXT5051_F700) Thanks __J

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[alsa-devel] Reg: Audio recording with twl4030 codec
by shariff 22 Jan '10

22 Jan '10

Hi all, I am working with OMAP3 based custom board that has twl4030 codec & using linux-2.6.29 kernel from opensource. I have some queries regarding Audio recording. Audio codec is connected to twl4030 as bellow TWL4030 |--------------------| | |------ MAINMIC McBSP2 ------ |i2s/tdm | | ___________| --> One Analog MIC is connected to Main Mic and I am using Audio path not the Voice path. After selecting Main Mic through amixer control. and recording audio using "arecord -c 1 -r 8000 -f S16_LE -t wav -d 20 record-file.wav" --> Here record is successful but with lot of noise. 1) Can we record Mono(1 channel) through I2S from TWL4030 codec? 2) To record stereo(2 channels), do we need two Mic's ? (i.e do I need Sub Mic also). Regards, Shariff

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Re: [alsa-devel] [PATCH 1/4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 22 Jan '10

22 Jan '10

On Tue, 19 Jan 2010, Mark Brown wrote: > On Tue, Jan 19, 2010 at 09:08:57AM +0100, Guennadi Liakhovetski wrote: > > This all looks pretty good - there's a few issues below but they're > largely stylistic rather than anything fundamental. > > > + 0x0000, 0x0000, 0x0000, 0x01ff, /* 0x08...0x0b */ /* 0x0b contains the VU bit */ > > + 0x01ff, 0x0000, 0x0100, 0x01ff, /* 0x0c...0x0f */ /* 0x0c and 0x0f contain the VU bit */ > > + 0x01ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ /* 0x10 contains the VU bit */ > > What do these comments refer to - do you mean to say that these are not > the actual chip register defaults? The normal way to deal with those is > to have a write (or other update of the cache defaults). This avoids > potential confusion later on if the chip updates the register defaults > or when reviewing against the datasheet. No, I meant, that there's a "Value Update" bit in that register, usually bit 8. So, that bit, of course, is not in default, but it was suggested to me on IRC to set it in cache to force automatic value update. > > +#define ARRAY_SINGLE(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ > > + ARRAY_SIZE(xtexts), xtexts) > > This looks generic - please namespace it and add it to the header file, > other drivers can benefit from it. SOC_ARRAY_SINGLE()? > > +static const struct soc_enum wm8978_enum[] = { > > + /* adc */ > > + ARRAY_SINGLE(WM8978_COMPANDING_CONTROL, 1, wm8978_companding), > > Please define individual variables for this - indexing into the array > gets hard to follow, it's far too easy to have an off by one errors > trying to match things up. > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > > > +/* Mixer #3: Boost (Input) mixer */ > > +static const struct snd_kcontrol_new wm8978_left_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Left PGA Mute", WM8978_LEFT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > +static const struct snd_kcontrol_new wm8978_right_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Right PGA Mute", WM8978_RIGHT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > These should have "Switch" at the end of the name. Also, the names are > going be concatenated with the mixer names so you'll end up with things > "Left Boost Mixer Left PGA Switch" - the individual controls should > probably be just "Switch" so you end up with "Left Boost Mixer Switch". > > > +/* Mic Input boost vol */ > > +static const struct snd_kcontrol_new wm8978_mic_boost_controls[] = { > > +SOC_DAPM_SINGLE("Left Mic Volume", WM8978_LEFT_ADC_BOOST_CONTROL, 4, 7, 0), > > +SOC_DAPM_SINGLE("Right Mic Volume", WM8978_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0), > > +}; > > This doesn't seem to be referenced anywhere? > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > Again, this should be done somewhere generic. Probably by going through > and changing the SND_SOC_DAPM_MIXER definition. Well, having discussed this privately and having looked at the heades, I'm not sure, this is a good idea. First, there are SND_SOC_DAPM_PGA() SND_SOC_DAPM_MIXER() SND_SOC_DAPM_MIXER_NAMED_CTL() SND_SOC_DAPM_PGA_E() SND_SOC_DAPM_MIXER_E() SND_SOC_DAPM_MIXER_NAMED_CTL_E() - 6 macros that follow the pattern .kcontrols = wcontrols, .num_kcontrols = wncontrols, and lots of macros like SOC_ENUM_DOUBLE() SOC_ENUM_SINGLE() SOC_ENUM_SINGLE_EXT() ... doing .max = xmax. BTW, many of the users even open-code array sizes instead of using ARRAY_SIZE() like static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", "Off"}; static const char *spk_function[] = {"On", "Off"}; static const struct soc_enum tosa_enum[] = { SOC_ENUM_SINGLE_EXT(5, jack_function), SOC_ENUM_SINGLE_EXT(2, spk_function), }; Changing only one of them would make the headers and the API look inconsistent. Changing all of them... I really don't think we want to endeavor this now, at least not me;) Besides, doing this locally is very convenient, and is available to everyone - there's no space science involved with this. One can be even as evil as doing +#define _A(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ + m, ARRAY_SIZE(m)) ... +#undef _A +#define _A(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ + ARRAY_SIZE(xtexts), xtexts) ... +#undef _A I think, it's best to leave these things as they are. Otherwise we can give this to kernel janitors as a small exercise;) > > + /* Output PLL to GPIO1 */ > > + snd_soc_write(codec, WM8978_GPIO_CONTROL, > > + snd_soc_read(codec, WM8978_GPIO_CONTROL) | 0x4); > > This should be being done unconditionally. Put a switch in via the > set_dai_clkdiv() call. Ok, I moved it to wm8978_set_dai_clkdiv() under "case WM8978_OPCLKDIV:" > > + /* Mic bias */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + (snd_soc_read(codec, 1) & ~4) | 0x10); > > + > > + /* Out-1 enabled, left/right input channel enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x1bf); > > + > > + /* Out-2 disabled, right/left output channel enabled, dac enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0x10f); > > These should all be being managed via DAPM. Yes, I'm having quite a bit of trouble with DAPM, but I'm getting through slowly. Will need some more time for a next version. > > > +static int wm8978_set_bias_level(struct snd_soc_codec *codec, > > + enum snd_soc_bias_level level) > > +{ > > + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; > > This bitmask maintains everything except the two LSB... > > > + switch (level) { > > + case SND_SOC_BIAS_ON: > > + case SND_SOC_BIAS_PREPARE: > > + power1 |= 1; /* VMID 75k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); > > + break; > > + case SND_SOC_BIAS_STANDBY: > > + power1 |= 0xC; > > ...but this is also managing other bits. Right, bits 7-6 are either kept or set, nothing wrong with that. > > + if (codec->bias_level == SND_SOC_BIAS_OFF) { > > + /* Initial cap charge at VMID 5k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + power1 | 0x3); > > + mdelay(100); > > + } > > > +/* Also supports 12kHz */ > > +#define WM8978_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ > > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | \ > > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) > > SNDRV_PCM_RATE_8000_48000 > > > +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) > > +{ > > + struct snd_soc_device *socdev = platform_get_drvdata(pdev); > > + struct snd_soc_codec *codec = socdev->card->codec; > > + > > + /* we only need to suspend if we are a valid card */ > > + if (!codec->card) > > + return 0; > > Don't need to check for the card any more, the core will stop you > getting called without a card. > > > + /* Sync reg_cache with the hardware */ > > + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { > > + if (i == WM8978_RESET) > > + continue; > > You can also skip the write if the register has the default value (this > will speed up resume slightly). > > > + data = cpu_to_be16((i << 9) | (cache[i] & 0x1ff)); > > + codec->hw_write(codec->control_data, (char *)&data, 2); > > + } > > Just use snd_soc_write()? I'll take care about the rest. Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Add PCI quirk for 17" MacBook Pro 5, 2 model (July 2009) to use existing mb5 model
by murray.alex＠gmail.com 22 Jan '10

22 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 7e0ea37..d044db9 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -9223,6 +9223,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x10ec, 0x0885, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- 1.6.3.3

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[alsa-devel] Update information: HD-Audio-models list
by Mark Brown 22 Jan '10

22 Jan '10

Sir/Madam, I have an update for the HD-Audio-Models.txt file. Under ALC262, the toshiba-s06 model setting is applicable to the Sony Vaio VGN-CS60B laptop, and may also be applicable to similar models. It also needs configuring in alsamixer to enable the internal microphone afterwards. Best regards, Mark Brown.

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Re: [alsa-devel] Reforming s3c2443 ac97 driver
by SeungHyun Choi 22 Jan '10

22 Jan '10

3 2

[alsa-devel] [PATCH] ASoC: add helper macros to declare struct soc_enum instances
by Guennadi Liakhovetski 21 Jan '10

21 Jan '10

Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..9537144 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -169,6 +169,23 @@ .private_value = (unsigned long)&xenum } /* + * Simplified versions of above macros, declaring a struct and calculating + * ARRAY_SIZE internally + */ +#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \ + struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \ + ARRAY_SIZE(xtexts), xtexts) +#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \ + SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts) +#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \ + struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts) +#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ + struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations.

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[alsa-devel] [PATCH 2/5] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 21 Jan '10

21 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 3 + sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 ++++ 4 files changed, 305 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..47e7cce 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..03f3feb --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,272 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

1 0

[alsa-devel] Problem with PPH plugin
by Colin Guthrie 21 Jan '10

21 Jan '10

Hi, Just wanted to highlight a bug over on Mozilla relating to a crash in the PPH plugin. https://bugzilla.mozilla.org/show_bug.cgi?id=540460 It was initially though to be in PulseAudio, but on closer inspection this doesn't involve PA at all and in fact seems to relate to the PPH plugin from what I can gather. The backtrace is here: https://bug540460.bugzilla.mozilla.org/attachment.cgi?id=422264 The user did submit a bug report to the bugtracker (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4878) but given previous experience there (see other mail!) I figured I'd post a message here to highlight it seeing as the reporter has already done a lot to try and work out what is going on and posting a message on his behalf seemed like a good idea :) Cheers All the best. -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

3 9

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 21 Jan '10

21 Jan '10

The following changes since commit 9135f6db09aa9d186f05d41e9c516ec8e3550d78: Mark Brown (1): Merge branch 'mxc-audio' into for-2.6.34 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Guennadi Liakhovetski (1): ASoC: fix compile breakage - add a missing header include Mark Brown (3): ASoC: Remove console DAPM debug code ASoC: Support turning off bias when the CODEC is idle ASoC: Use BIAS_OFF when idle for wm_hubs devices Peter Ujfalusi (3): ASoC: tlv320dac33: Correct the prefill number of samples ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/soc-dai.h | 2 + include/sound/soc.h | 2 + include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 24 +++++++-- sound/soc/codecs/wm_hubs.c | 6 ++ sound/soc/soc-dapm.c | 105 ++++++++++---------------------------- 6 files changed, 56 insertions(+), 84 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Support turning off bias when the CODEC is idle
by Mark Brown 21 Jan '10

21 Jan '10

Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc.h | 2 ++ sound/soc/soc-dapm.c | 25 ++++++++++++++++++++++++- 2 files changed, 26 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 08909cc..a8768ea 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -405,6 +405,8 @@ struct snd_soc_codec { short reg_cache_size; short reg_cache_step; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + /* dapm */ u32 pop_time; struct list_head dapm_widgets; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d8e9374..6c33510 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1012,13 +1012,28 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) sys_power = 0; break; case SND_SOC_DAPM_STREAM_NOP: - sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + sys_power = 0; + break; + default: + sys_power = 1; + break; + } break; default: break; } } + if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_STANDBY); + if (ret != 0) + pr_err("Failed to turn on bias: %d\n", ret); + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -1042,6 +1057,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply standby bias: %d\n", ret); } + /* If we're in standby and can support bias off then do that */ + if (codec->bias_level == SND_SOC_BIAS_STANDBY && + codec->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + if (ret != 0) + pr_err("Failed to turn off bias: %d\n", ret); + } + /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { ret = snd_soc_dapm_set_bias_level(socdev, -- 1.6.6

1 0

[alsa-devel] Reforming s3c2443 ac97 driver
by JASWINDERSINGH BRAR 21 Jan '10

21 Jan '10

Hi, Right now the S3C2443's AC97_ver2.0 controller driver isn't flexible enough to handle newer SoCs. In order to prepare ground for supporting 64xx and S5Pxxxx the extant driver needs to be reformed, and that is what i am planning to do next. I plan to do the following:- 1) User platform_driver structure to detect and manage AC97 controller platform devices. 2) Remove hardcoded parameters(DMA, IO address, IRQ etc) and pass via platform resources. 3) Initialize platform specific stuff by callback pointers to platform code(cfg_gpio) Objections, suggestions welcome. ------------------------------- Jaswinder Singh Brar Senior Engineer Linux OS Team Application Processor Development System LSI Division SAMSUNG Electronics Co., Ltd. --------------------------------

5 8

[alsa-devel] snd_pcm_hw_params_free() immediately after snd_pcm_hw_params() ???
by Raymond Yau 21 Jan '10

21 Jan '10

Refer to http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/ch05s06.html#pcm-… hw_free callback static int snd_xxx_hw_free(struct snd_pcm_substream *substream); This is called to release the resources allocated via *hw_params*. For example, releasing the buffer via snd_pcm_lib_malloc_pages() is done by calling the following: snd_pcm_lib_free_pages(substream); This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track whether the resource was already released. Nearly all the examples in those ALSA tutorials in the web sites have the code if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)); exit (1); } snd_pcm_hw_params_free (hw_params); This seem to violate "This function is always called before the close callback is called." and those hardware mixing sound cards (e.g cs46xx , au88x0, ... ) use snd_xxx_hw_free() to deallocate the hardware mixing path in the DSP. There are a number of audio package (e.g. gstreamer , wine , .... ) call snd_pcm_hw_params_free immediately after snd_pcm_hw_params() Is this a bug of snd_pcm_hw_params_free() not change the state of the PCM stream ? or snd_pcm_prepare() did not check the buffer is already released ?

2 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA850/OMAP-L138 EVM
by Chaithrika U S 21 Jan '10

21 Jan '10

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

3 3

[alsa-devel] device naming
by Andrew Kohlsmith (Mailing List Account) 20 Jan '10

20 Jan '10

Is there a definitive guide to device naming with ALSA? I'm trying to use a single device for simultaneous capture and playback but opening different device names gives different results and I'm trying to determine which is the best to use. aplay -L output gives front, surround40, surround41, surround50, surround51, null and pulse. Trying to register a playback callback using "pulse" gives a "function not implemented" error, although I can open and set all the parameters I need (8kHz, 2 channels, 20ms buffers). I get the same result with "default". If I use "plughw:0,0" I can capture and play audio, but the driver (intel HDA) seems to be very prone to either not calling either the playback or capture callback, or throwing an xrun and not being able to recover. This is an otherwise idle core2 machine running Ubuntu 9.10. Playing with nice or ionice does not seem to affect this "brittleness." I do not have an ~/.asoundrc, nor do I have anything related to asound in /etc. Moving pulseaudio out of the way doesn't seem to do anything either. Is there a definitive list of devices and their device filenames so I can stop guessing? /proc/asound/cards just shows card 0 as an Intel HDA, and /proc/asound/devices shows the following: $ cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0- 1]: hardware dependent 8: [ 0- 0]: hardware dependent 9: [ 0] : control (0-0 is the only device showing both playback and capture, which is why I'm using it, but how do I know that plughw is the correct device filename? I just guessed at it due to previous experience trying to get bt audio working with ALSA.) I'm sure I'm doing something incredibly stupid, but I can't seem to find out what it is. -A.

1 0

[alsa-devel] [PATCH 2/2] Davinci: DM365: Enable DaVinci Voice Codec support for DM365 EVM
by miguel.aguilar＠ridgerun.com 20 Jan '10

20 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> It sets up one single card for the DM365 EVM with two subdevices: the AIC3x and the Voice Codec. The general structures are defined at DM365 SoC file and the specific platform data structure for the EVM is defined at board file. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 39 +++++++++++++++++++++++++++- arch/arm/mach-davinci/dm365.c | 37 ++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 1 + arch/arm/mach-davinci/include/mach/dm365.h | 5 +++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 5 files changed, 80 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 8d23972..74f3519 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -175,7 +175,43 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; -static struct snd_platform_data dm365_evm_snd_data; +static int dm365_snd_dma_event_mux(int device, int enable, int playback) +{ + static int tx_dma_mux = 0; + static int rx_dma_mux = 0; + + if (!enable) { + if (playback) + tx_dma_mux = 0; + else + rx_dma_mux = 0; + return 0; + } + + if (playback) { + if (tx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT2_ASP_TX); + else + davinci_cfg_reg(DM365_EVT2_VC_TX); + tx_dma_mux = 1; + } else { + if (rx_dma_mux == 1) + return -EBUSY; + if (device == 0) + davinci_cfg_reg(DM365_EVT3_ASP_RX); + else + davinci_cfg_reg(DM365_EVT3_VC_RX); + rx_dma_mux = 1; + } + + return 0; +} + +static struct snd_platform_data dm365_evm_snd_data = { + .dma_event_mux = &dm365_snd_dma_event_mux, +}; static struct i2c_board_info i2c_info[] = { { @@ -513,6 +549,7 @@ static __init void dm365_evm_init(void) evm_init_cpld(); dm365_init_asp(&dm365_evm_snd_data); + dm365_init_vc(&dm365_evm_snd_data); dm365_init_rtc(); #ifdef CONFIG_KEYBOARD_DAVINCI diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index cc3bae4..024cbab 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct davinci_clk dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -604,6 +604,8 @@ INT_CFG(DM365, INT_IMX1_DISABLE, 24, 1, 0, false) INT_CFG(DM365, INT_NSF_ENABLE, 25, 1, 1, false) INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) #endif @@ -835,6 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -985,12 +1012,16 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) davinci_cfg_reg(DM365_MCBSP0_BDR); davinci_cfg_reg(DM365_MCBSP0_R); davinci_cfg_reg(DM365_MCBSP0_BFSR); - davinci_cfg_reg(DM365_EVT2_ASP_TX); - davinci_cfg_reg(DM365_EVT3_ASP_RX); dm365_asp_device.dev.platform_data = pdata; platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { davinci_cfg_reg(DM365_KEYSCAN); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..6a175eb 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -50,6 +50,7 @@ #define DAVINCI_ASP1_TX_INT IRQ_MBXINT struct snd_platform_data { + int (*dma_event_mux)(int, int, int); u32 tx_dma_offset; u32 rx_dma_offset; enum dma_event_q eventq_no; /* event queue number */ diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..50aeaec 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index b60c693..2937c51 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -325,6 +325,8 @@ enum davinci_dm365_index { DM365_INT_NSF_DISABLE, /* EDMA event muxing */ + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, DM365_EVT26_MMC0_RX, -- 1.6.0.4

4 13

[alsa-devel] [PATCH] Obsolete config in kernel source (USE_MIXARTLOADER)
by Christoph Egger 20 Jan '10

20 Jan '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we're checking referential integrity between kernel KConfig options and in-code Conditional blocks. The config Variable USE_MIXARTLOADER has been non-existent for all git based linux kernels (2.6.12-rc) and seems to be no longer relevant. It is only referenced at one place in the Code (sound/pci/mixart/mixart_hwdep.c). The attached patch makes the behaviour more obvious as it is equivalent to the old one when assuming CONFIG_MIXARTLOADER as not defined. Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christoph Egger

2 3

[alsa-devel] Conexant CX20561 on HP Pavilion dv2700
by Jacopo De Simoi 20 Jan '10

20 Jan '10

Dear alsa-devs I'm trying to have CX20561 in working order on my laptop; I'll be glad to send a patch once I figure out what is going on there, and for this I need your help. There are a number of issues, mostly regarding the internal mic, as I'm sure you know considering the bugs volume on the subject :) 1a) There are three capture sources listed in alsamixer (Digital excluded): Docking External Internal; the only one that has any influence in recording (both from internal and external mics) is "External" 1b) Why does Node 0x14 shows two volume controls? the first one doesn't seem to have any effect. 2) The input from the internal mic device has very low volume and is very noisy; I tried to optimize fiddling around with hda_verb and the values in alsa-info; the output of codec#0 below shows the optimal ones. (the only relevant ones were gains on node 0x17). I managed, however, to saturate it by blowing directly into it, which shows that it is not really an amplification issue. Rather the real issue here is noise; Audacity shows ~-24dB and in 16bits the noise background gets up to ~+/-500 versus +/-50 with an external mic. The internal mic is indeed a stereo mic; the left channel only produces noise, the right channel seems to work, with the caveats above. (this could be very well that my mic is broken, even if I remember that at some point during my fiddling with hda_verb it started to work properly :) in any case I noticed that the noise background is very much correlated between R and L; would it be possible to write an alsa-plugin to mix the two channels with an inversion to mix away the noise? It sounds fairly simple, but I couldn't find any documentation on how to make a plugin I'm willing to invest some time on fixing this, so maybe I can get some help here to start with. Judging from the patch_conexant code it should be at least possible to fix #1. I'd like to hear your thoughts about #2; especially about fiddling around with hda_verb or making the described alsa-plugin. Thanks a lot Jacopo De Simoi !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Wed Jan 20 08:49:43 UTC 2010 !!Linux Distribution !!------------------ Gentoo Base System release 2.0.0 !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP Pavilion dv2700 Notebook PC !!Kernel Information !!------------------ Kernel release: 2.6.30-tuxonice-r9 Operating System: GNU/Linux Architecture: x86_64 Processor: AMD Turion(tm) 64 X2 TL-62 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22.1 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfc480000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:07.0 0403: 10de:055c (rev a1) Subsystem: 103c:30d6 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=laptop snd-hda-intel: model=auto !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : laptop,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Conexant CX20561 (Hermosa) Address: 0 Function Id: 0x2 Vendor Id: 0x14f15051 Subsystem Id: 0x103c30d6 Revision Id: 0x100000 Modem Function Group: 0x2 Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x3b 0x3b] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=0 Amp-Out vals: [0x4a 0x4a] Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Internal Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Internal Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="External Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="External Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Device: name="CONEXANT Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] [0x50 0x50] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x1d 0x17* Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Docking Mic Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Docking Mic Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=0 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01214030: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=37, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=38, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001224: IN Detect Vref caps: 50 80 Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=39, enabled=1 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1a [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1b [Pin Complex] wcaps 0x400500: Mono Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400701: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="Conexant Digital", type="SPDIF", device=1 Pincap 0x00000010: OUT Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x1d [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-In vals: [0x04 0x04] Pincap 0x00000020: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x1e [Vendor Defined Widget] wcaps 0xf00000: Mono --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jan 19 20:26 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jan 19 20:26 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 24 Jan 20 09:31 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 20 09:34 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 17 Jan 19 20:26 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 1 Jan 19 20:26 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jan 19 20:26 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfc480000 irq 20' Mixer name : 'Conexant CX20561 (Hermosa)' Components : 'HDA:14f15051,103c30d6,00100000' Controls : 15 Simple ctrls : 8 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 59 [80%] [-15.00dB] [off] Front Right: Playback 59 [80%] [-15.00dB] [off] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 0 [0%] [-30.00dB] Front Right: Capture 0 [0%] [-30.00dB] Simple mixer control 'Docking Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] Simple mixer control 'External Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 80 [100%] [6.00dB] Playback [on] Front Right: 80 [100%] [6.00dB] Playback [on] Simple mixer control 'Internal Mic',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 80 Front Left: 0 [0%] [-74.00dB] Playback [on] Front Right: 0 [0%] [-74.00dB] Playback [on] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Internal Mic Volume' value.0 0 value.1 0 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Internal Mic Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'External Mic Volume' value.0 80 value.1 80 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'External Mic Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 80' comment.dbmin -7400 comment.dbmax 600 iface MIXER name 'Docking Mic Volume' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Docking Mic Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 74' comment.dbmin -7400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 59 value.1 59 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value false } control.9 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.10 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.11 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.14 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } control.15 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module ndiswrapper bnep sco rfcomm l2cap snd_seq snd_seq_device snd_hda_codec_conexant uvcvideo videodev snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_timer sdhci_pci sdhci mmc_core v4l1_compat nvidia forcedeth snd btusb bluetooth v4l2_compat_ioctl32 led_class snd_page_alloc !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x16 0x01214030 0x17 0x01a19020 0x18 0x400001f0 0x19 0x400001f0 0x1a 0x92170110 0x1b 0x400001f0 0x1c 0x400001f0 0x1d 0x400001f0 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 8.457898] ACPI: PCI Interrupt Link [LAZA] enabled at IRQ 20 [ 8.457916] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [ 8.457970] HDA Intel 0000:00:07.0: setting latency timer to 64 [ 8.499562] Linux video capture interface: v2.00 -- [ 8.548267] USB Video Class driver (v0.1.0) [ 9.660207] input: HDA NVidia Mic as /class/input/input10 [ 9.660575] input: HDA NVidia Mic as /class/input/input11 [ 9.660798] input: HDA NVidia Headphone as /class/input/input12 [ 10.842906] kjournald starting. Commit interval 5 seconds -- [18156.018398] forcedeth 0000:00:0a.0: PCI INT A disabled [18156.400042] HDA Intel 0000:00:07.0: PCI INT A disabled [18156.440030] ehci_hcd 0000:00:04.1: PCI INT B disabled -- [18157.038988] ehci_hcd 0000:00:04.1: restoring config space at offset 0x1 (was 0xb00006, writing 0xb00002) [18157.039014] HDA Intel 0000:00:07.0: restoring config space at offset 0xf (was 0x5020100, writing 0x502010a) [18157.039022] HDA Intel 0000:00:07.0: restoring config space at offset 0x4 (was 0x0, writing 0xfc480000) [18157.039027] HDA Intel 0000:00:07.0: restoring config space at offset 0x1 (was 0xb00000, writing 0xb00002) [18157.039057] ahci 0000:00:09.0: restoring config space at offset 0x1 (was 0xb00007, writing 0xb00407) -- [18157.150089] amd74xx 0000:00:06.0: BIOS didn't set cable bits correctly. Enabling workaround. [18157.150111] HDA Intel 0000:00:07.0: PCI INT A -> Link[LAZA] -> GSI 20 (level, low) -> IRQ 20 [18157.150115] HDA Intel 0000:00:07.0: setting latency timer to 64 [18157.850034] pci 0000:00:08.0: setting latency timer to 64

1 1

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --+pHx0qQiF2pBVqBT--

1 0

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --liOOAslEiF7prFVr--

1 0

[PATCH] config USE_MIXARTLOADER has never existed in KConfig for 2.6 linux kernels so this part of the check always evaluates to true thus removing the part of the check.
by Christoph Egger 20 Jan '10

20 Jan '10

The same is true for USE_PCXHRLOADER and USE_VXLOADER Signed-off-by: Christoph Egger <siccegge(a)stud.informatik.uni-erlangen.de> --- include/sound/vx_core.h | 2 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5456343..c6835dd 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -28,7 +28,7 @@ #include <linux/interrupt.h> #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_VXLOADER) && !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ +#if !defined(CONFIG_SND_VX_LIB) /* built-in kernel */ #define SND_VX_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..ef39a58 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -545,7 +545,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_MIXARTLOADER) && !defined(CONFIG_SND_MIXART) /* built-in kernel */ +#if !defined(CONFIG_SND_MIXART) /* built-in kernel */ #define SND_MIXART_FW_LOADER /* use the standard firmware loader */ #endif #endif diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 17cb123..16152ce 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -35,7 +35,7 @@ #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) -#if !defined(CONFIG_USE_PCXHRLOADER) && !defined(CONFIG_SND_PCXHR) /* built-in kernel */ +#if !defined(CONFIG_SND_PCXHR) /* built-in kernel */ #define SND_PCXHR_FW_LOADER /* use the standard firmware loader */ #endif #endif -- 1.6.3.3 --bCsyhTFzCvuiizWE--

1 0

Re: [alsa-devel] [PATCH] ASoC: DaVinci: Fix stream restart error
by Chaithrika U S 20 Jan '10

20 Jan '10

All, Please ignore this patch, missed the 'To' address in the mail. I have re-submitted this patch. Regards, Chaithrika On Wed, Jan 20, 2010 at 17:04:45, alsa-devel(a)alsa-project.org wrote: > From: Chaithrika U S <chaithrika(a)ti.com> > > Sometimes after a suspend-resume cycle, the ALSA application > restarts the stream when resume fails and McASP fails to work > as the clock is not enabled. This patch corrects this bug. > > Testes on TI DA850/OMAP-L138 EVM. > > Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> > --- > Applies to ALSA GIT tree on branch topic/asoc at > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; > h=topic/asoc > > sound/soc/davinci/davinci-mcasp.c | 5 ++--- > 1 files changed, 2 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c > index a613bbb..ab6518d 100644 > --- a/sound/soc/davinci/davinci-mcasp.c > +++ b/sound/soc/davinci/davinci-mcasp.c > @@ -768,13 +768,12 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, > > switch (cmd) { > case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > if (!dev->clk_active) { > clk_enable(dev->clk); > dev->clk_active = 1; > } > - /* Fall through */ > - case SNDRV_PCM_TRIGGER_START: > - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > davinci_mcasp_start(dev, substream->stream); > break; > > -- > 1.5.6 >

1 0

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: Burst mode BCLK divider fixes
by Peter Ujfalusi 20 Jan '10

20 Jan '10

Hello, The following patches provides fixes for the BCLK clock divider configuration in tlv320dac33 codec. BCLK divider is used to specify the speed of the serial bus. The speed of the bus can be (must be) faster than the actual stream's sample rate in order to use the system in burst mode. With the introduction of mode7 FIFO mode, the BCLK handling got broken in this mode, the first patch fixes this issue. The second patch adds support for configuring the burst mode BCLK through platform data. --- Peter Ujfalusi (2): ASoC: tlv320dac33: BCLK divider fix ASoC: tlv320dac33: Burst mode BCLK divider configuration include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 22 +++++++++++++++++----- 2 files changed, 18 insertions(+), 5 deletions(-)

3 4

[alsa-devel] Can we add optional parameters in snd_pcm_hw_params
by Raul Xiong 20 Jan '10

20 Jan '10

Hi all, In my current ALSA driver, I have defined a mode which will impact the configuration of the audio driver. The requirment is to ask application users to set this mode to driver. So I want to add a parameter in 'hwparams' of snd_pcm_hw_params(pcm_handle, hwparams). But it seems that ALSA have no such expansibility. Basically, my requirment is: There are two modes: ModeA and ModeB. I'd like application to set this mode before it begin playing. If you have other better methods to reach this requirment, it's also welcome. Thanks.

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[alsa-devel] State of the bugtracker
by Colin Guthrie 20 Jan '10

20 Jan '10

Hi, I remember a while back I asked about the bugtracker over at https://bugtrack.alsa-project.org/alsa-bug/my_view_page.php At the time it was generally regarded that the bugtracker was a bit of a black hole. I've generally avoided using it to report bugs, but I'd like to ask what it's current state is as I often have to direct users to *something* and generally go for the mailing list. So really this is just a "what is the state of the bug tracker" question. Do the main devs check it and triage it or is there no specific process? Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

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[alsa-devel] No alsa-driver snapshots since 2009-12-19?
by Aleksander Adamowski 20 Jan '10

20 Jan '10

Hi! Seems that alsa-driver snapshots have stopped being updated on http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ - the last one is from 2009-12-19. What happened? -- Best Regards, Aleksander Adamowski http://olo.org.pl

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[alsa-devel] [PATCH] Fix automute issue on HP T5735
by Lee Trager 20 Jan '10

20 Jan '10

The following patch fixes an issue where when you plug something into the headphone jack the internal speaker does not automute. This also will close alsa bug #0004766 Lee --- alsa-driver-orig/alsa-kernel/pci/hda/patch_realtek.c 2009-08-28 07:41:39.000000000 -0400 +++ alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-10-13 14:24:40.000000000 -0400 @@ -10028,7 +10028,9 @@ struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; + + alc_automute_amp(codec); } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {

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Re: [alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by Miguel Aguilar 19 Jan '10

19 Jan '10

Hi Mark, I have been trying to test the AIC3x and the Voice Codec in the same kernel, however only the AIC3x is working, I'm having problem when I try to use the Voice Codec, I already have verified that the Voice Codec works since I tested alone. Please check the following output: root@dm365-evm:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EVM [DaVinci DM365 EVM], device 0: AIC3X tlv320aic3x-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: EVM [DaVinci DM365 EVM], device 1: CQ93 CQ93VC-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@dm365-evm:~# aplay -L default:CARD=EVM DaVinci DM365 EVM, Default Audio Device null Discard all samples (playback) or generate zero samples (capture) root@dm365-evm:~# cat /proc/asound/pcm 00-00: AIC3X tlv320aic3x-0 : : playback 1 : capture 1 00-01: CQ93 CQ93VC-1 : : playback 1 : capture 1 root@dm365-evm:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 33: : timer root@dm365-evm:~# cat /proc/asound/cards 0 [EVM ]: tlv320aic3x - DaVinci DM365 EVM DaVinci DM365 EVM (tlv320aic3x) root@dm365-evm:~# cat /proc/asound/timers G0: system timer : 10000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-1-0: PCM playback 0-1-0 : SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE root@dm365-evm:~# arecord -D plughw:0,0 test.wav Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono *** This is the error that I have. I not sure if it has to do with the root@dm365-evm:~# arecord -D plughw:0,1 test.wav asoc: can't open platform davinci-audio arecord: main:608: audio open error: No such device The following is the patch to the davinci-evm.c please check it if you see something wrong there: diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..951d730 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -29,8 +29,10 @@ #include "../codecs/tlv320aic3x.h" #include "../codecs/spdif_transciever.h" +#include "../codecs/cq93vc.h" #include "davinci-pcm.h" #include "davinci-i2s.h" +#include "davinci-vcif.h" #include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ @@ -151,6 +153,23 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, + }, +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +196,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +204,14 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = ARRAY_SIZE(dm365_evm_dai), +}; + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +244,13 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +278,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; Thanks, Miguel Aguilar

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[alsa-devel] [PATCH] ASoC: fix compile breakage - add a missing header include
by Guennadi Liakhovetski 19 Jan '10

19 Jan '10

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- It might well have been reported and / or fixed in some trees, sorry for the noise, if that's the case. include/sound/soc-dai.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ca24e7f..061f16d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -16,6 +16,8 @@ #include <linux/list.h> +#include <sound/soc.h> + struct snd_pcm_substream; /* -- 1.6.2.4

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Correct the prefill number of samples
by Peter Ujfalusi 19 Jan '10

19 Jan '10

Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2df9c20..65683aa 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -559,7 +559,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) break; case DAC33_FIFO_MODE7: dac33_write16(codec, DAC33_PREFILL_MSB, - DAC33_THRREG(20)); + DAC33_THRREG(10)); break; default: dev_warn(codec->dev, "Unhandled FIFO mode: %d\n", -- 1.6.6

2 1

[alsa-devel] About audio driver architecture
by Michael Qiu 19 Jan '10

19 Jan '10

Hi, I'm going to write an audio driver for a developping soc used for dvb set top box. As I've never done such things before, so I've some questions. 1. Seems the "ALSA Driver Documentation" is mainly for PCI(and for PC) based audio card. Where can I got a document for SOC which has no PCI bus. If there are some existing drivers it will be even better. 2. Is there any standard for such a audio driver which upper applications will depend on it? Or I can just follow ALSA? Best regards Michael Qiu

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Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 19 Jan '10

19 Jan '10

2010/1/16 Lars Zobbe <fedesensfar(a)gmail.com> > Thansk a lot Raymond. That actually works! My HDSP now loads without mayor > problems. This is brilliant. I've been struggling for so long to make this > work and finally... > I recommend everyone with a RME HDSP, who wants to use pulseaudio, to check > out ticket 764. > Thanks again Raymond. > Lars. > > 2010/1/14 Raymond Yau <superquad.vortex2(a)gmail.com> > >> >> Refer to >> >> http://pulseaudio.org/ticket/764#comment:1 >> >> >> load-module module-alsa-sink device_id=0 tsched=0 >> > load-module module-alsa-source device_id=0 tsched=0 >> > if your hdsp works , this mean that this is a bug introduced at pulseaudio since 0.9.15 ( working profile ) I guess none of PA develpers had your card , if you want the pulseaudio bug fixed , you will need to post the output of full pulseaudio log of module-alsa-sink ( pulseaudio -k;pulseaudio -vvvvv ) and the outout of aplay -v -Dplughw:0,0 any.wav in the pulseaudio bug tracker so that PA developer can compare with the non-working pulseaudio log

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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: Voice Codec Support
by miguel.aguilar＠ridgerun.com 18 Jan '10

18 Jan '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> There are two clients for the MFD driver: * Voice Codec Interface. * Voice Codec itself, CQ0093. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 5 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 216 ++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 113 +++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 342 ++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/davinci/Kconfig | 10 +- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.h | 28 +++ 11 files changed, 750 insertions(+), 2 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..efeba26 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,11 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + bool "DaVinci Voice Codec" + help + This driver supports the DaVinci voice codec and its interface. + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..94dbeaa --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,216 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/core.h> +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int davinci_vc_client_dev_register(struct davinci_vc *davinci_vc, + const char *name, + struct platform_device **pdev) +{ + int ret; + + *pdev = platform_device_alloc(name, -1); + if (pdev == NULL) { + dev_err(davinci_vc->dev, "failed to allocate %s\n", name); + return -ENODEV; + } + + (*pdev)->dev.parent = davinci_vc->dev; + platform_set_drvdata(*pdev, davinci_vc); + ret = platform_device_add(*pdev); + if (ret != 0) { + dev_err(davinci_vc->dev, "failed to register %s: %d\n", name, + ret); + platform_device_put(*pdev); + *pdev = NULL; + return ret; + } + + return 0; +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + + /* Register voice codec interface */ + ret = davinci_vc_client_dev_register(davinci_vc, "davinci_vcif", + &(davinci_vc->davinci_vcif.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail4; + } + + /* Register voice codec */ + ret = davinci_vc_client_dev_register(davinci_vc, "cq93vc", + &(davinci_vc->cq93vc.pdev)); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client device\n"); + goto fail5; + } + + return 0; + +fail5: + platform_device_unregister(davinci_vc->davinci_vcif.pdev); +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + platform_device_unregister(davinci_vc->cq93vc.pdev); + platform_device_unregister(davinci_vc->davinci_vcif.pdev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..6a1757a --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,113 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..a759d06 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..674db53 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..f420e01 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,342 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(cq93vc_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&cq93vc_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Set the PGA Gain to 18 dB */ + cq93vc_write(codec, DAVINCI_VC_REG05, DAVINCI_VC_REG05_PGA_GAIN); + + /* Set the DAC digital attenuation to 0 dB */ + cq93vc_write(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_DIG_ATTEN); + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + cq93vc_add_controls(codec); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + /* Power down chip */ + if (codec->control_data) + cq93vc_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .suspend = cq93vc_suspend, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int cq93vc_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int cq93vc_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define cq93vc_codec_suspend NULL +#define cq93vc_codec_resume NULL +#endif + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), + .suspend = cq93vc_codec_suspend, + .resume = cq93vc_codec_resume, +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..3aa04e6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,15 +12,21 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

3 3

[alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 18 Jan '10

18 Jan '10

Hello Fellows, Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. Do you know what I might have to do to get it working on Gentoo (sabayon)? When I type the following, this is what I get: ~/Downloads/alsa-utils-1.0.22/alsactl $ ./alsactl init 2 Unknown hardware: "Echo_Indigo IO" "" "" "0xecc0" "0x00a0" Hardware is initialized using a guess method Using aplay to list the device: card 2: IO [Indigo IO], device 0: PCM [Indigo IO] Subdevices: 8/8 .... etc. Using aplay to test the card here is the command: aplay -D plughw:2,0 <wav file> And this does not work, I have checked that things are ok with alsamixer, but still! Any help will be appreciated or let me know if this is best addressed by submitting a bug. // Salvador

2 2

[alsa-devel] WARNING: at sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]()
by John Kacur 18 Jan '10

18 Jan '10

The following was in my 2.6.33-rc3 dmesg (of course that doesn't mean that the problem didn't exist earlier too) WARNING: at /home/jkacur/rt.linux.git/sound/core/sound_oss.c:96 snd_oss_kernel_minor+0xdf/0x120 [snd]() Hardware name: 2241B48 BUG? (minor < 0 || minor >= 128) Modules linked in: snd_mixer_oss thinkpad_acpi(+) sdhci hwmon i2c_i801 firewire_ohci snd_pcm mmc_core snd_timer cfg80211 firewire_core btusb joydev iTCO_wdt bluetooth ata_generic e1000e i2c_core ppdev snd battery ac iTCO_vendor_support parport_pc ricoh_mmc intel_agp pata_acpi pcspkr parport tpm_tis video wmi output soundcore button tpm rfkill sr_mod tpm_bios snd_page_alloc crc_itu_t cdrom sg ahci libata sd_mod scsi_mod crc_t10dif xfs exportfs uhci_hcd ohci_hcd ehci_hcd [last unloaded: scsi_wait_scan] Pid: 1317, comm: modprobe Not tainted 2.6.33-rc3 #1 Call Trace: [<ffffffffa027bb25>] ? snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffff8104338c>] warn_slowpath_common+0x7c/0xa9 [<ffffffff81043410>] warn_slowpath_fmt+0x41/0x43 [<ffffffffa027bb25>] snd_oss_kernel_minor+0xdf/0x120 [snd] [<ffffffffa027bc7e>] snd_register_oss_device+0x2c/0x21a [snd] [<ffffffffa0271e1e>] snd_mixer_oss_notify_handler+0xa5/0x2e5 [snd_mixer_oss] [<ffffffff81375e00>] ? __mutex_unlock_slowpath+0x5c/0x141 [<ffffffff810710d5>] ? trace_hardirqs_on_caller+0x11f/0x14a [<ffffffff8107110d>] ? trace_hardirqs_on+0xd/0xf sdhci-pci 0000:15:00.2: SDHCI controller found [1180:0822] (rev 21) sdhci-pci 0000:15:00.2: PCI INT C -> GSI 18 (level, low) -> IRQ 18 [<ffffffff81375ed3>] ? __mutex_unlock_slowpath+0x12f/0x141 sdhci-pci 0000:15:00.2: Will use DMA mode even though HW doesn't fully claim to support it. Registered led device: mmc0:: mmc0: SDHCI controller on PCI [0000:15:00.2] using DMA [<ffffffffa027628a>] snd_card_register+0x135/0x14b [snd] [<ffffffffa0370c88>] volume_init+0x335/0x3d2 [thinkpad_acpi] [<ffffffffa037289b>] thinkpad_acpi_module_init+0x61f/0xa4f [thinkpad_acpi] [<ffffffffa037227c>] ? thinkpad_acpi_module_init+0x0/0xa4f [thinkpad_acpi] [<ffffffff810001fa>] do_one_initcall+0x5f/0x154 [<ffffffff8107ff34>] sys_init_module+0xd7/0x234 [<ffffffff81002d5b>] system_call_fastpath+0x16/0x1b ---[ end trace a6d53dc318f09cb5 ]---

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[alsa-devel] issue with sound stopping and not being able to restart
by Andrew Kohlsmith (Mailing List Account) 18 Jan '10

18 Jan '10

Good morning, I am new to ALSA development, but experienced in C. I wrote up a little test application which simply opens one playback and one capture device and spits noise into the playback and recording it on the capture device. I cross- correlate the data streams to determine the "depth" or latency of the pipe, end-to-end. I'm using playback and capture callbacks to handle the audio. The program seems to work reasonably well, but at seemingly random times either the playback or capture stream will "hiccup" (xrun) -- I am handling this case, but it's fairly often that the stream does NOT resume and I'm stuck with either no playback or no capture. I've looked at the example programs and the documentation and I do not see what it is I'm doing wrong. I'm posting my code here, hopefully someone can show me the error of my way. Capture/playback setup (error checking removed): snd_pcm_open(handle, fname, stream, SND_PCM_NONBLOCK); snd_pcm_hw_params_malloc(hwptr); snd_pcm_hw_params_any(h, hwp); snd_pcm_hw_params_set_access(h, hwp, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(h, hwp, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate(h, hwp, 8000, 0); snd_pcm_hw_params_set_channels(h, hwp, 2); snd_pcm_hw_params_set_buffer_size(h, hwp, BUF_SIZE); snd_pcm_hw_params(h, hwp); At this point I have the capture or playback device set up for 16 bit signed linear data in interleaved format and two channels. I've set up the buffers to be 160 frames, or 20ms. Each of these calls is checked for errors, and I do verify that the rate, buffer size and buffer time are what I expect. There is nothing unusual going on with these calls that I can determine. I set up the playback and capture callbacks thusly (again checking for errors) snd_async_add_pcm_handler(&ahp, hp, playback_callback, &pb_data); snd_async_add_pcm_handler(&ahc, hc, capture_callback, &cap_data); At this point I believe I'm ready to go, all I have to do is start playback and capture: if((fcount = snd_pcm_writei(h[PLAYBACK], pb_data.buf, BUF_SIZE)) != BUF_SIZE) { if(fcount < 0) { if((e = snd_pcm_recover(h[PLAYBACK], fcount, 0)) < 0) fprintf(stderr, "initial: can't writei: %s\n", snd_strerror(e)); } else { fprintf(stderr, "initial: short write: %d instead of %d\n", fcount, BUF_SIZE); } } if((e = snd_pcm_start(h[CAPTURE])) < 0) { fprintf(stderr, "capture prepare error: %s\n", snd_strerror(e)); return 1; } Now the playback and capture handlers are called by ALSA as required. The capture handler looks like this: /* capture callback. Called by ALSA whenever a period time is up */ void capture_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; short *p; int i, fc, e, faloops; struct timeval tv; static int c=0; /* read in whatever is available */ fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; if((fc = snd_pcm_readi(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't readi: %s\n", snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short read: %d instead of %ld\n", fc, fa); } } for(i=0,p=&d->buf[0]; i<fc; i++) { *d->ptr++ = *p++; p++; if(d->ptr > &in[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&in[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; in_step++; if(in_step >= MAX_BUFS) in_step = 0; } frames_cap += fc; fa = snd_pcm_avail_update(h); faloops++; }; cap_faloops=faloops; } The playback callback is very similar: /* playback callback. Called by ALSA whenever a period time is up */ void playback_callback(snd_async_handler_t *ah) { snd_pcm_t *h = snd_async_handler_get_pcm(ah); struct async_data *d = snd_async_handler_get_callback_private(ah); snd_pcm_sframes_t fa; int i, fc, e, faloops; static int c=0; struct timeval tv; fa = snd_pcm_avail_update(h); faloops = 0; while(fa > 0) { if(fa > BUF_SIZE) fa = BUF_SIZE; for(i=0; i<fa; i++) { d->buf[i*2] = awgn(&noise_source); d->buf[i*2+1] = awgn(&noise_source); } if((fc = snd_pcm_writei(h, d->buf, fa)) != fa) { if(fc < 0) { if((e = snd_pcm_recover(h, e, 0)) < 0) { fprintf(stderr, "callback: can't writei: error %x %s\n", e, snd_strerror(e)); } fc = 0; } else { fprintf(stderr, "callback: short write: %d instead of %ld\n", fc, fa); } } for(i=0; i<fc; i++) { *d->ptr++ = d->buf[i*2]; if(d->ptr > &out[MAX_BUFS * (BUF_SIZE - 1)]) { d->ptr = (short *)&out[0]; } } c += fc; if(c >= BUF_SIZE) { gettimeofday(&tv, NULL); c -= BUF_SIZE; out_step++; if(out_step >= MAX_BUFS) out_step = 0; } frames_pb += fc; fa = snd_pcm_avail_update(h); faloops++; }; pb_faloops = faloops; } That's really it; the main loop keeps track of the in_step and out_step variables and does the cross-correlation as the data comes in. Now it seems that it's the capture playback that is getting xruns, but I when capture xruns occur my playback callback seems to stop, and sometimes the entire program just hangs as if it is waiting for ALSA to call one of the callbacks. Can anyone shed any light on this? I'm certain it's something very simple and braindead, but I can't find it. Regards, Andrew

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 18 Jan '10

18 Jan '10

> Actually i am working on soc platform(android-S3C6410) and wanted to > use DMIX and some good resampler for my software mixing instead of > current software mixing and resampling in upper layer(Above > alsa-lib).DMIX is working fine but this resampling is causing some > jitter.During rate conversion it looks for > libasound_module_rate_speexrate.so which i don't have so its using > default one causing the jitter. >>Have you considered using PulseAudio for this? Mark thanks for the suggestion and i checked out PulseAudio for my requirements.I found out that i don't need all the things in PulseAudio a. I just need good resampling code.Can i just use just the resampler code of pulseaudio(mine is mobile enviroment)? b. Just want a good mixer at ALSA-LIB level.As of now this is provided by android framework but i want this to be done by ALSA-LIB so that i can replace with some good mixer later on(if i find that some other mixer is good).DMIX is providing that but it's not that good. c.Jaroslav suggested to use sox resamplers but for that to work i have to get the sox resampler code(proper license) and need to build it for my system and link it with corresponding functions of ALSA-LIB so that they can call relevant functions for resampling.As i am working on mobile domain i think i need to do the integration of sox resampler code with ALSA-LIB for that to work?? or Can i just use the integrated sox library(some .so i guess) and change the asound.conf for using that library??Will this work?? d. I saw some check-ins related to Scenario Manager.As of now scenario manager functionality is also given in Android(we open a differnet device and the corresponding hooks for changing the routes dynamically) but i think if it given in alsa-lib switching will be much faster and this will increase the system performance.

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[alsa-devel] [PATCH] add atomic operations for Blackfin parts
by Mike Frysinger 18 Jan '10

18 Jan '10

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- include/iatomic.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 8f6ec22..e92dbfd 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1079,6 +1079,29 @@ static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) #endif /* __sh__ */ +#ifdef __bfin__ + +#include <bfin_fixed_code.h> + +typedef struct { volatile int counter; } atomic_t; + +#define ATOMIC_INIT(i) { (i) } + +#define atomic_read(v) ((v)->counter) +#define atomic_set(v,i) (((v)->counter) = (i)) +#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) +#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) +#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); +#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); + +#define mb() __asm__ __volatile__ ("" : : : "memory") +#define rmb() mb() +#define wmb() mb() + +#define IATOMIC_DEFINED 1 + +#endif /* __bfin__ */ + #ifndef IATOMIC_DEFINED /* * non supported architecture. -- 1.6.6

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Jan '10

18 Jan '10

The following changes since commit 735fe4cfbc3cedea41bd0ed31955054dae6beb46: Mark Brown (1): ASoC: Add missing __devexit and __devinit annotations are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (3): ASoC: ak4104: allow more sample rates ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() ASoC: support more sample rates on raumfeld devices Mark Brown (6): ASoC: Fix i.MX audio build for i.MX3x ASoC: Convert new i.MX SSI driver to use static DAI array ASoC: i.MX SSI driver does not yet support master mode ASoC: Remove old i.MX driver code ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Merge branch 'mxc-audio' into for-2.6.34 Peter Ujfalusi (1): ASoC: TWL4030: Replace comma with semicolon in probe function Sascha Hauer (1): ASoC: Add a new imx-ssi sound driver Thomas Weber (1): ASoC: TWL4030: Fix typo in comment in header file arch/arm/plat-mxc/Makefile | 4 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 ++++++ sound/soc/codecs/ak4104.c | 4 +- sound/soc/codecs/cs4270.c | 38 ++- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl4030.h | 2 +- sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++ sound/soc/imx/imx-ssi.c | 758 +++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 237 +++++++++++ sound/soc/imx/mx1_mx2-pcm.c | 488 --------------------- sound/soc/imx/mx1_mx2-pcm.h | 26 -- sound/soc/imx/mx27vis_wm8974.c | 318 -------------- sound/soc/imx/mxc-ssi.c | 860 -------------------------------------- sound/soc/imx/mxc-ssi.h | 238 ----------- sound/soc/pxa/raumfeld.c | 61 ++- 19 files changed, 1823 insertions(+), 1989 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h delete mode 100644 sound/soc/imx/mx1_mx2-pcm.c delete mode 100644 sound/soc/imx/mx1_mx2-pcm.h delete mode 100644 sound/soc/imx/mx27vis_wm8974.c delete mode 100644 sound/soc/imx/mxc-ssi.c delete mode 100644 sound/soc/imx/mxc-ssi.h

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[alsa-devel] MSM DSP
by GNUtoo 17 Jan '10

17 Jan '10

hi, I've bought an HTC dream,before buying I looked at the feasibility of porting GNU/Linux on it(so not Android). There is a patch(for maing the driver standard) or a standard(everything is non-standard in android) driver for most of the components. The biggest problem seems the sound. So knowing that codeaurora( https://www.codeaurora.org/gitweb/quic/le/?p=kernel/msm.git ) had alsa in its kenrel I first tried to compile that kernel...but due to numerous bugs(It didn't compile and needed too much fixing) in it android-msm-2.6.29 branch I decided to import alsa in the linuxtogo kernel,and I pushed the sources at a temporary htc-alsa branch at git://gitorious.org/replicant/gnulinuxkernel.git (it's called gnulinux in order to differentiate from an android kernel,I didn't find a good name for it) from the commit message available here : http://gitorious.org/replicant/gnulinuxkernel/commit/92945ccd0921e940fc2675… I had to change snd_rpc_ids.vers = 0x00020001 into snd_rpc_ids.vers = 0xaa2b1a44 ,and to make minor build system changes and I had aplay working rapidely... but only aplay... mplayer was waiting indefinitely for the following system call: ioctl(6, 0x400c4150 ls -l /proc/$(pido mplayer)/fd/6 gave /dev/snd/pcmC0D0p and 0x400c4150 gave that: SNDRV_PCM_IOCTL_WRITEI_FRAMES = _IOW('A', 0x50, struct snd_xferi) Then with printks(I had no serial cable(I've looked for a craddle/docking station with the special HTC connector for easy soldering...but I didn't found one) so I couldn't use kgdb) I went until that point: rc = wait_event_interruptible(the_locks.write_wait, (frame->used == 0)|| (prtd->stopped)); That's in the alsa_send_buffer function in /sound/soc/msm/msm-pcm.c Then I continued the tracing puting printks everywhere...and I found that: with aplay: msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was called rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was called once inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) the count loop was only executed once because count was exhausted on the first run then copy_count is incremented and the audio is configured then at second call of the alsa_send_buffer the audio is configured and works. here's an output of the aplay parameters: $ aplay -v 05\ -\ Jet\ Set.wav Playing WAVE '05 - Jet Set.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'msm-audio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4800 period_size : 1200 period_time : 27210 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1200 xfer_align : 1200 start_threshold : 4800 stop_threshold : 4800 silence_threshold: 0 silence_size : 0 boundary : 1258291200 note that frame = prtd->out + prtd->out_tail; makes frame->used have a bad value the first time it is run but as the count is only run once it doesn't import. the next runs prtd->out + prtd->out_tail have a good value with mplayer the story was quite different,even if the beginning was the same : msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was also called,rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was also called inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) but once in alsa_send_buffer,more precisely under the count loop the story changes: count is not exhausted the first time,because in "xfer = count > frame->size ? frame->size : count;" the smallest between count and frame->size is assigned,and so after it does "count -= xfer" and so count is not 0,because the buffer size detected by mplayer is 19200 insterad of 4800 so here what it does in detail: it passes 19200 as count here: fbytes = frames_to_bytes(runtime, frames); rc = alsa_send_buffer(prtd, buf, fbytes, NULL); that goes into that: ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, size_t count, loff_t *pos) and then the count variable is 19200 in alsa_send_buffer then there is xfer = count > frame->size ? frame->size : count; so it assigns the minimum between count and frame->size to xfer count is still 19200 frame->size is 4800 so xfer becomes 4800 then count -= xfer decrement count but only of 4800 so count is not exhausted and runs another time then : frame = prtd->out + prtd->out_tail; makes frame->used have a bad value,and then it blocks here: rc = wait_event_interruptible(the_locks.write_wait,(frame->used == 0) || (prtd->stopped)); because frame->used is not 0. here an mplayer output: Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options '[none]' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.15 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opened in blocking mode alsa-init: got buffersize=19200 alsa-init: got period size 300 alsa: 44100 Hz/2 channels/4 bpf/19200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under developement Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 21248 so the quick solution was to force mplayer to use 4800 as buffer size with something like that: diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c index 38e8283..bfbc66d 100644 --- a/sound/soc/msm/msm7k-pcm.c +++ b/sound/soc/msm/msm7k-pcm.c @@ -115,7 +115,7 @@ static struct snd_pcm_hardware msm_pcm_playback_hardware = { .rate_max = USE_RATE_MAX, .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, - .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE, + .buffer_bytes_max = 4800, .period_bytes_min = 64, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, but then mplayer seem to loop on an ioctl: mplayer is frozen here and doesn't respond to other input than ctrl+c: 0.0 (00.0) of 206.0 (03:26.0) ??,?% strace shows a lot of : nanosleep followed by SNDRV_PCM_IOCTL_STATUS (ioctl(6, 0x806c4120, 0xbe8bc9c8) = 0) an analyse of the status shows a constant : status->delay is 1200,and status->avail is 0,they don't vary over time and are printed just after that code in snd_pcm_status in sound/core/pcm_native.c if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || runtime->status->state == SNDRV_PCM_STATE_DRAINING) status->delay = runtime->buffer_size - status->avail; else status->delay = 0; I was told in #mplayerdev that it was because the DSP didn't play the frames...indeed I hear no sound.... So how should I fix that mess? *did the quick and dirty fix cause the second problem? *should I use the DSP directly and make a pulseaudio plugin that talks to the android driver, instead(the driver seems very buggy) The good would be support for mp3 and similar things within the DSP,and that it should be more stable. The bad is that I already spent *a lot* of time with the alsa kernel driver...and that writing that will also require some userspaces changes such as: *modification of the phone routing in the "FSO" ( http://www.freesmartphone.org/index.php/Main_Page ) dbus framework to use pulseaudio *would require changes in the build system for using pulseaudio(openembedded(http://wiki.openembedded.net/index.php/Main_Page )) *may be slow...openmoko stopped to use pulseaudio because of speed issues. Denis

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[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

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