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[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.

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[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.

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[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

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[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

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[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

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[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

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