From 87af38dafe4f930921b9217c21ec6d72cad56bcc Mon Sep 17 00:00:00 2001 From: Daniel Jacobowitz Date: Wed, 7 May 2008 12:05:10 +0200 Subject: [ALSA] ac97 - Add virtual master control to VT1616/VT1617A codec. Enable VMASTER for VT1616 / VT1617A codec. Signed-off-by: Daniel Jacobowitz Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 76 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 76 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1292dcee072d..92817f7d46d2 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3352,8 +3352,66 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0), AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0), }; +static const char *slave_vols_vt1616[] = { + "Front Playback Volume", + "Surround Playback Volume", + "Center Playback Volume", + "LFE Playback Volume", + NULL +}; + +static const char *slave_sws_vt1616[] = { + "Front Playback Switch", + "Surround Playback Switch", + "Center Playback Switch", + "LFE Playback Switch", + NULL +}; + +/* find a mixer control element with the given name */ +static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97, + const char *name) +{ + struct snd_ctl_elem_id id; + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, name); + return snd_ctl_find_id(ac97->bus->card, &id); +} + +/* create a virtual master control and add slaves */ +int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name, + const unsigned int *tlv, const char **slaves) +{ + struct snd_kcontrol *kctl; + const char **s; + int err; + + kctl = snd_ctl_make_virtual_master(name, tlv); + if (!kctl) + return -ENOMEM; + err = snd_ctl_add(ac97->bus->card, kctl); + if (err < 0) + return err; + + for (s = slaves; *s; s++) { + struct snd_kcontrol *sctl; + + sctl = snd_ac97_find_mixer_ctl(ac97, *s); + if (!sctl) { + snd_printdd("Cannot find slave %s, skipped\n", *s); + continue; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + } + return 0; +} + static int patch_vt1616_specific(struct snd_ac97 * ac97) { + struct snd_kcontrol *kctl; int err; if (snd_ac97_try_bit(ac97, 0x5a, 9)) @@ -3361,6 +3419,24 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97) return err; if ((err = patch_build_controls(ac97, &snd_ac97_controls_vt1616[1], ARRAY_SIZE(snd_ac97_controls_vt1616) - 1)) < 0) return err; + + /* There is already a misnamed master switch. Rename it. */ + kctl = snd_ac97_find_mixer_ctl(ac97, "Master Playback Volume"); + if (!kctl) + return -EINVAL; + + snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback"); + + err = snd_ac97_add_vmaster(ac97, "Master Playback Volume", + kctl->tlv.p, slave_vols_vt1616); + if (err < 0) + return err; + + err = snd_ac97_add_vmaster(ac97, "Master Playback Switch", + NULL, slave_sws_vt1616); + if (err < 0) + return err; + return 0; } -- cgit v1.2.3 From d023dc0aa25d6a4f7dd0d109179a2077d22a7ad2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:18:27 +0200 Subject: [ALSA] oxygen: fix version in MODULE_LICENSE Adjust the MODULE_LICENSE strings to properly reflect the actual license. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/oxygen_lib.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 090dd4354a28..c5d2c86cacb2 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -28,7 +28,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("TempoTec HiFier driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 63f185c1ed1e..c1b30cc75935 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -43,7 +43,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8788}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 897697d43506..6e3879527dcd 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -32,7 +32,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); static irqreturn_t oxygen_interrupt(int dummy, void *dev_id) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 7f84fa5deca2..4618a622a296 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -79,7 +79,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("Asus AVx00 driver"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; -- cgit v1.2.3 From c13650079ba3bed1c0bdd9bf4a13274be7676ff6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:19:53 +0200 Subject: [ALSA] oxygen: add symbol for I/O space size Remove another magic number - add a symbol for the size of the PCI I/O range. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen.h | 2 ++ sound/pci/oxygen/oxygen_lib.c | 4 ++-- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index a71c6e059260..6c6efede4d2f 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -16,6 +16,8 @@ #define PCM_AC97 5 #define PCM_COUNT 6 +#define OXYGEN_IO_SIZE 0x100 + /* model-specific configuration of outputs/inputs */ #define PLAYBACK_0_TO_I2S 0x001 #define PLAYBACK_1_TO_SPDIF 0x004 diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 6e3879527dcd..60bc15917349 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -173,7 +173,7 @@ static void oxygen_proc_read(struct snd_info_entry *entry, int i, j; snd_iprintf(buffer, "CMI8788\n\n"); - for (i = 0; i < 0x100; i += 0x10) { + for (i = 0; i < OXYGEN_IO_SIZE; i += 0x10) { snd_iprintf(buffer, "%02x:", i); for (j = 0; j < 0x10; ++j) snd_iprintf(buffer, " %02x", oxygen_read8(chip, i + j)); @@ -455,7 +455,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } if (!(pci_resource_flags(pci, 0) & IORESOURCE_IO) || - pci_resource_len(pci, 0) < 0x100) { + pci_resource_len(pci, 0) < OXYGEN_IO_SIZE) { snd_printk(KERN_ERR "invalid PCI I/O range\n"); err = -ENXIO; goto err_pci_regions; -- cgit v1.2.3 From e58aee95806c9d2bbcfc84cb85ce958e360165ef Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:20:51 +0200 Subject: [ALSA] oxygen: save register writes Save the written values of all CMI8788 and AC97 registers and of some of the DAC/ADC registers so that it is possible to restore the register state later. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen.c | 17 +++++++++++--- sound/pci/oxygen/oxygen.h | 6 +++++ sound/pci/oxygen/oxygen_io.c | 22 ++++++++++++++---- sound/pci/oxygen/virtuoso.c | 54 ++++++++++++++++++++++++-------------------- 4 files changed, 67 insertions(+), 32 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index c1b30cc75935..8287ac28fe78 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -80,6 +80,7 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); struct generic_data { u8 ak4396_ctl2; + u16 saved_wm8785_registers[2]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -99,12 +100,16 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) { + struct generic_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); + if (reg < ARRAY_SIZE(data->saved_wm8785_registers)) + data->saved_wm8785_registers[reg] = value; } static void ak4396_init(struct oxygen *chip) @@ -135,10 +140,16 @@ static void ak5385_init(struct oxygen *chip) static void wm8785_init(struct oxygen *chip) { + struct generic_data *data = chip->model_data; + + data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | + WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->saved_wm8785_registers[1] = WM8785_WL_24; + wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST); - wm8785_write(chip, WM8785_R1, WM8785_WL_24); + wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); + wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); + snd_component_add(chip->card, "WM8785"); } diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6c6efede4d2f..29c9fa4964b2 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -80,6 +80,12 @@ struct oxygen { struct work_struct spdif_input_bits_work; struct work_struct gpio_work; wait_queue_head_t ac97_waitqueue; + union { + u8 _8[OXYGEN_IO_SIZE]; + __le16 _16[OXYGEN_IO_SIZE / 2]; + __le32 _32[OXYGEN_IO_SIZE / 4]; + } saved_registers; + u16 saved_ac97_registers[2][0x40]; }; struct oxygen_model { diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 5569606ee87f..83f135f80df4 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -44,18 +44,21 @@ EXPORT_SYMBOL(oxygen_read32); void oxygen_write8(struct oxygen *chip, unsigned int reg, u8 value) { outb(value, chip->addr + reg); + chip->saved_registers._8[reg] = value; } EXPORT_SYMBOL(oxygen_write8); void oxygen_write16(struct oxygen *chip, unsigned int reg, u16 value) { outw(value, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(value); } EXPORT_SYMBOL(oxygen_write16); void oxygen_write32(struct oxygen *chip, unsigned int reg, u32 value) { outl(value, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(value); } EXPORT_SYMBOL(oxygen_write32); @@ -63,7 +66,10 @@ void oxygen_write8_masked(struct oxygen *chip, unsigned int reg, u8 value, u8 mask) { u8 tmp = inb(chip->addr + reg); - outb((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outb(tmp, chip->addr + reg); + chip->saved_registers._8[reg] = tmp; } EXPORT_SYMBOL(oxygen_write8_masked); @@ -71,7 +77,10 @@ void oxygen_write16_masked(struct oxygen *chip, unsigned int reg, u16 value, u16 mask) { u16 tmp = inw(chip->addr + reg); - outw((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outw(tmp, chip->addr + reg); + chip->saved_registers._16[reg / 2] = cpu_to_le16(tmp); } EXPORT_SYMBOL(oxygen_write16_masked); @@ -79,7 +88,10 @@ void oxygen_write32_masked(struct oxygen *chip, unsigned int reg, u32 value, u32 mask) { u32 tmp = inl(chip->addr + reg); - outl((tmp & ~mask) | (value & mask), chip->addr + reg); + tmp &= ~mask; + tmp |= value & mask; + outl(tmp, chip->addr + reg); + chip->saved_registers._32[reg / 4] = cpu_to_le32(tmp); } EXPORT_SYMBOL(oxygen_write32_masked); @@ -128,8 +140,10 @@ void oxygen_write_ac97(struct oxygen *chip, unsigned int codec, oxygen_write32(chip, OXYGEN_AC97_REGS, reg); /* require two "completed" writes, just to be sure */ if (oxygen_ac97_wait(chip, OXYGEN_AC97_INT_WRITE_DONE) >= 0 && - ++succeeded >= 2) + ++succeeded >= 2) { + chip->saved_ac97_registers[codec][index / 2] = data; return; + } } snd_printk(KERN_ERR "AC'97 write timeout\n"); } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 4618a622a296..110786b2e3b8 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -132,6 +132,9 @@ struct xonar_data { u8 ext_power_int_reg; u8 ext_power_bit; u8 has_power; + u8 pcm1796_oversampling; + u8 cs4398_fm; + u8 cs4362a_fm; }; static void pcm1796_write(struct oxygen *chip, unsigned int codec, @@ -186,12 +189,13 @@ static void xonar_d2_init(struct oxygen *chip) data->anti_pop_delay = 300; data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->pcm1796_oversampling = PCM1796_OS_64; for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, PCM1796_OS_64); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); pcm1796_write(chip, i, 21, 0); pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ pcm1796_write(chip, i, 17, 0x0f); @@ -226,6 +230,9 @@ static void xonar_dx_init(struct oxygen *chip) data->ext_power_reg = OXYGEN_GPI_DATA; data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->ext_power_bit = GPI_DX_EXT_POWER; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | @@ -236,8 +243,7 @@ static void xonar_dx_init(struct oxygen *chip) cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); /* configure */ - cs4398_write(chip, 2, CS4398_FM_SINGLE | - CS4398_DEM_NONE | CS4398_DIF_LJUST); + cs4398_write(chip, 2, data->cs4398_fm); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); cs4398_write(chip, 5, 0xfe); @@ -249,16 +255,13 @@ static void xonar_dx_init(struct oxygen *chip) CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x06, data->cs4362a_fm); cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x09, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x09, data->cs4362a_fm); cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0c, CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); /* clear power down */ @@ -294,12 +297,13 @@ static void xonar_dx_cleanup(struct oxygen *chip) static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { + struct xonar_data *data = chip->model_data; unsigned int i; - u8 value; - value = params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; + data->pcm1796_oversampling = + params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 20, value); + pcm1796_write(chip, i, 20, data->pcm1796_oversampling); } static void update_pcm1796_volume(struct oxygen *chip) @@ -342,24 +346,24 @@ static void set_cs53x1_params(struct oxygen *chip, static void set_cs43xx_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { - u8 fm_cs4398, fm_cs4362a; + struct xonar_data *data = chip->model_data; - fm_cs4398 = CS4398_DEM_NONE | CS4398_DIF_LJUST; - fm_cs4362a = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; if (params_rate(params) <= 50000) { - fm_cs4398 |= CS4398_FM_SINGLE; - fm_cs4362a |= CS4362A_FM_SINGLE; + data->cs4398_fm |= CS4398_FM_SINGLE; + data->cs4362a_fm |= CS4362A_FM_SINGLE; } else if (params_rate(params) <= 100000) { - fm_cs4398 |= CS4398_FM_DOUBLE; - fm_cs4362a |= CS4362A_FM_DOUBLE; + data->cs4398_fm |= CS4398_FM_DOUBLE; + data->cs4362a_fm |= CS4362A_FM_DOUBLE; } else { - fm_cs4398 |= CS4398_FM_QUAD; - fm_cs4362a |= CS4362A_FM_QUAD; + data->cs4398_fm |= CS4398_FM_QUAD; + data->cs4362a_fm |= CS4362A_FM_QUAD; } - cs4398_write(chip, 2, fm_cs4398); - cs4362a_write(chip, 0x06, fm_cs4362a); - cs4362a_write(chip, 0x09, fm_cs4362a); - cs4362a_write(chip, 0x0c, fm_cs4362a); + cs4398_write(chip, 2, data->cs4398_fm); + cs4362a_write(chip, 0x06, data->cs4362a_fm); + cs4362a_write(chip, 0x09, data->cs4362a_fm); + cs4362a_write(chip, 0x0c, data->cs4362a_fm); } static void update_cs4362a_volumes(struct oxygen *chip) -- cgit v1.2.3 From bbbfb5526650cb9d01c5c230a4e3cbbc18474c41 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:21:48 +0200 Subject: [ALSA] oxygen: simplify DAC volume initialization When initializing the DAC volume registers, we can just use the generic volume update functions instead of setting the registers manually. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 15 +++--- sound/pci/oxygen/oxygen.c | 27 +++++----- sound/pci/oxygen/virtuoso.c | 123 ++++++++++++++++++++------------------------ 3 files changed, 77 insertions(+), 88 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index c5d2c86cacb2..0e425d554505 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -62,6 +62,12 @@ static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) AK4396_WRITE | (reg << 8) | value); } +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); +} + static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; @@ -70,8 +76,7 @@ static void hifier_init(struct oxygen *chip) ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, AK4396_LCH_ATT, 0); - ak4396_write(chip, AK4396_RCH_ATT, 0); + update_ak4396_volume(chip); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); @@ -100,12 +105,6 @@ static void set_ak4396_params(struct oxygen *chip, ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); } -static void update_ak4396_volume(struct oxygen *chip) -{ - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); -} - static void update_ak4396_mute(struct oxygen *chip) { struct hifier_data *data = chip->model_data; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 8287ac28fe78..ad8d950fddc6 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -112,6 +112,18 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) data->saved_wm8785_registers[reg] = value; } +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, + AK4396_LCH_ATT, chip->dac_volume[i * 2]); + ak4396_write(chip, i, + AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); + } +} + static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; @@ -125,9 +137,8 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, i, AK4396_CONTROL_3, AK4396_PCM); - ak4396_write(chip, i, AK4396_LCH_ATT, 0); - ak4396_write(chip, i, AK4396_RCH_ATT, 0); } + update_ak4396_volume(chip); snd_component_add(chip->card, "AK4396"); } @@ -194,18 +205,6 @@ static void set_ak4396_params(struct oxygen *chip, } } -static void update_ak4396_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_LCH_ATT, chip->dac_volume[i * 2]); - ak4396_write(chip, i, - AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); - } -} - static void update_ak4396_mute(struct oxygen *chip) { struct generic_data *data = chip->model_data; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 110786b2e3b8..bd9e28583518 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -182,6 +182,28 @@ static void xonar_common_init(struct oxygen *chip) oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); } +static void update_pcm1796_volume(struct oxygen *chip) +{ + unsigned int i; + + for (i = 0; i < 4; ++i) { + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + unsigned int i; + u8 value; + + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < 4; ++i) + pcm1796_write(chip, i, 18, value); +} + static void xonar_d2_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -192,14 +214,12 @@ static void xonar_d2_init(struct oxygen *chip) data->pcm1796_oversampling = PCM1796_OS_64; for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 18, PCM1796_MUTE | PCM1796_DMF_DISABLED | - PCM1796_FMT_24_LJUST | PCM1796_ATLD); pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, data->pcm1796_oversampling); pcm1796_write(chip, i, 21, 0); - pcm1796_write(chip, i, 16, 0x0f); /* set ATL/ATR after ATLD */ - pcm1796_write(chip, i, 17, 0x0f); } + update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ + update_pcm1796_volume(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); @@ -221,6 +241,37 @@ static void xonar_d2x_init(struct oxygen *chip) xonar_d2_init(chip); } +static void update_cs4362a_volumes(struct oxygen *chip) +{ + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); + cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); + cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); + cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); + cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); + cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write(chip, 4, reg); + update_cs4362a_volumes(chip); +} + static void xonar_dx_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -245,9 +296,6 @@ static void xonar_dx_init(struct oxygen *chip) /* configure */ cs4398_write(chip, 2, data->cs4398_fm); cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); - cs4398_write(chip, 4, CS4398_MUTEP_LOW | CS4398_PAMUTE); - cs4398_write(chip, 5, 0xfe); - cs4398_write(chip, 6, 0xfe); cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); @@ -256,14 +304,10 @@ static void xonar_dx_init(struct oxygen *chip) cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); cs4362a_write(chip, 0x05, 0); cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x07, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x08, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0a, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0b, 0x7f | CS4362A_MUTE); cs4362a_write(chip, 0x0c, data->cs4362a_fm); - cs4362a_write(chip, 0x0d, 0x7f | CS4362A_MUTE); - cs4362a_write(chip, 0x0e, 0x7f | CS4362A_MUTE); + update_cs43xx_volume(chip); + update_cs43xx_mute(chip); /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); @@ -306,28 +350,6 @@ static void set_pcm1796_params(struct oxygen *chip, pcm1796_write(chip, i, 20, data->pcm1796_oversampling); } -static void update_pcm1796_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } -} - -static void update_pcm1796_mute(struct oxygen *chip) -{ - unsigned int i; - u8 value; - - value = PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; - for (i = 0; i < 4; ++i) - pcm1796_write(chip, i, 18, value); -} - static void set_cs53x1_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -366,37 +388,6 @@ static void set_cs43xx_params(struct oxygen *chip, cs4362a_write(chip, 0x0c, data->cs4362a_fm); } -static void update_cs4362a_volumes(struct oxygen *chip) -{ - u8 mute; - - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); -} - -static void update_cs43xx_volume(struct oxygen *chip) -{ - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); -} - -static void update_cs43xx_mute(struct oxygen *chip) -{ - u8 reg; - - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); -} - static void xonar_gpio_changed(struct oxygen *chip) { struct xonar_data *data = chip->model_data; -- cgit v1.2.3 From 75146fc0f9368ea41419792ac8bfdd19273b4473 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:22:43 +0200 Subject: [ALSA] oxygen: separate out hardware initialization code Create separate functions for the code that initializes the hardware, as opposed to initializing internal driver state, so that they can be reused for resume support. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 11 +++++++-- sound/pci/oxygen/oxygen.c | 26 ++++++++++++++++------ sound/pci/oxygen/virtuoso.c | 54 ++++++++++++++++++++++++++++----------------- 3 files changed, 62 insertions(+), 29 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 0e425d554505..6e45a58ad14b 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -68,15 +68,22 @@ static void update_ak4396_volume(struct oxygen *chip) ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } -static void hifier_init(struct oxygen *chip) +static void hifier_registers_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); update_ak4396_volume(chip); +} + +static void hifier_init(struct oxygen *chip) +{ + struct hifier_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + hifier_registers_init(chip); snd_component_add(chip->card, "AK4396"); snd_component_add(chip->card, "CS5340"); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ad8d950fddc6..800ae304a247 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -124,12 +124,11 @@ static void update_ak4396_volume(struct oxygen *chip) } } -static void ak4396_init(struct oxygen *chip) +static void ak4396_registers_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; unsigned int i; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); @@ -139,6 +138,14 @@ static void ak4396_init(struct oxygen *chip) AK4396_CONTROL_3, AK4396_PCM); } update_ak4396_volume(chip); +} + +static void ak4396_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + ak4396_registers_init(chip); snd_component_add(chip->card, "AK4396"); } @@ -149,18 +156,23 @@ static void ak5385_init(struct oxygen *chip) snd_component_add(chip->card, "AK5385"); } -static void wm8785_init(struct oxygen *chip) +static void wm8785_registers_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; - data->saved_wm8785_registers[1] = WM8785_WL_24; - wm8785_write(chip, WM8785_R7, 0); wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); +} +static void wm8785_init(struct oxygen *chip) +{ + struct generic_data *data = chip->model_data; + + data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | + WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->saved_wm8785_registers[1] = WM8785_WL_24; + wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bd9e28583518..e95dc5717ed7 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -204,15 +204,11 @@ static void update_pcm1796_mute(struct oxygen *chip) pcm1796_write(chip, i, 18, value); } -static void xonar_d2_init(struct oxygen *chip) +static void pcm1796_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; unsigned int i; - data->anti_pop_delay = 300; - data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - for (i = 0; i < 4; ++i) { pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); pcm1796_write(chip, i, 20, data->pcm1796_oversampling); @@ -220,6 +216,17 @@ static void xonar_d2_init(struct oxygen *chip) } update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ update_pcm1796_volume(chip); +} + +static void xonar_d2_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 300; + data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->pcm1796_oversampling = PCM1796_OS_64; + + pcm1796_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); @@ -272,24 +279,10 @@ static void update_cs43xx_mute(struct oxygen *chip) update_cs4362a_volumes(chip); } -static void xonar_dx_init(struct oxygen *chip) +static void cs43xx_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; - data->anti_pop_delay = 800; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - /* set CPEN (control port mode) and power down */ cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); @@ -311,6 +304,27 @@ static void xonar_dx_init(struct oxygen *chip) /* clear power down */ cs4398_write(chip, 8, CS4398_CPEN); cs4362a_write(chip, 0x01, CS4362A_CPEN); +} + +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->anti_pop_delay = 800; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4362a_fm = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + cs43xx_init(chip); oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); -- cgit v1.2.3 From 92215f3a178080bd9d7c65879499e9474e54d55c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:23:02 +0200 Subject: [ALSA] virtuoso: add xonar_enable_output() Move the setting of the output enable GPIO bit to a separate function. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/virtuoso.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index e95dc5717ed7..abd5313d19b5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -162,6 +162,14 @@ static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); } +static void xonar_enable_output(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + static void xonar_common_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -173,13 +181,12 @@ static void xonar_common_init(struct oxygen *chip) data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) & data->ext_power_bit); } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_CS53x1_M_MASK | data->output_enable_bit); oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - msleep(data->anti_pop_delay); - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); + xonar_enable_output(chip); } static void update_pcm1796_volume(struct oxygen *chip) -- cgit v1.2.3 From 4a4bc53bc52978dd6c918531921da925fd047d95 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:24:39 +0200 Subject: [ALSA] oxygen: add PM support Add suspend/resume support. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/hifier.c | 5 +++ sound/pci/oxygen/oxygen.c | 12 +++++ sound/pci/oxygen/oxygen.h | 6 +++ sound/pci/oxygen/oxygen_lib.c | 100 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/oxygen/oxygen_pcm.c | 1 + sound/pci/oxygen/virtuoso.c | 22 ++++++++++ 6 files changed, 146 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 6e45a58ad14b..7442460583dd 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -146,6 +146,7 @@ static const struct oxygen_model model_hifier = { .init = hifier_init, .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, + .resume = hifier_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, @@ -186,6 +187,10 @@ static struct pci_driver hifier_driver = { .id_table = hifier_ids, .probe = hifier_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_hifier_init(void) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 800ae304a247..7c8ae31eb468 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -192,6 +192,12 @@ static void generic_cleanup(struct oxygen *chip) { } +static void generic_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); + wm8785_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -278,6 +284,7 @@ static const struct oxygen_model model_generic = { .owner = THIS_MODULE, .init = generic_init, .cleanup = generic_cleanup, + .resume = generic_resume, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, @@ -305,6 +312,7 @@ static const struct oxygen_model model_meridian = { .owner = THIS_MODULE, .init = meridian_init, .cleanup = generic_cleanup, + .resume = ak4396_registers_init, .set_dac_params = set_ak4396_params, .set_adc_params = set_ak5385_params, .update_dac_volume = update_ak4396_volume, @@ -353,6 +361,10 @@ static struct pci_driver oxygen_driver = { .id_table = oxygen_ids, .probe = generic_oxygen_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_oxygen_init(void) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 29c9fa4964b2..74a644880074 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -97,6 +97,8 @@ struct oxygen_model { int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); void (*cleanup)(struct oxygen *chip); + void (*suspend)(struct oxygen *chip); + void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); void (*set_dac_params)(struct oxygen *chip, @@ -125,6 +127,10 @@ struct oxygen_model { int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, const struct oxygen_model *model); void oxygen_pci_remove(struct pci_dev *pci); +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); +int oxygen_pci_resume(struct pci_dev *pci); +#endif /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 60bc15917349..22f37851045e 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -314,6 +314,10 @@ static void oxygen_init(struct oxygen *chip) OXYGEN_SPDIF_LOCK_MASK | OXYGEN_SPDIF_RATE_MASK); oxygen_write32(chip, OXYGEN_SPDIF_OUTPUT_BITS, chip->spdif_bits); + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_STANDARD); oxygen_clear_bits8(chip, OXYGEN_MPU401_CONTROL, OXYGEN_MPU401_LOOPBACK); oxygen_write8(chip, OXYGEN_GPI_INTERRUPT_MASK, 0); oxygen_write16(chip, OXYGEN_GPIO_INTERRUPT_MASK, 0); @@ -534,3 +538,99 @@ void oxygen_pci_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } EXPORT_SYMBOL(oxygen_pci_remove); + +#ifdef CONFIG_PM +int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i, saved_interrupt_mask; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + for (i = 0; i < PCM_COUNT; ++i) + if (chip->streams[i]) + snd_pcm_suspend(chip->streams[i]); + + if (chip->model->suspend) + chip->model->suspend(chip); + + spin_lock_irq(&chip->reg_lock); + saved_interrupt_mask = chip->interrupt_mask; + chip->interrupt_mask = 0; + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + spin_unlock_irq(&chip->reg_lock); + + synchronize_irq(chip->irq); + flush_scheduled_work(); + chip->interrupt_mask = saved_interrupt_mask; + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_suspend); + +static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = { + 0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff, + 0x00300000, 0x00000fe4, 0x0ff7001f, 0x00000000 +}; +static const u32 ac97_registers_to_restore[2][0x40 / 32] = { + { 0x18284fa2, 0x03060000 }, + { 0x00007fa6, 0x00200000 } +}; + +static inline int is_bit_set(const u32 *bitmap, unsigned int bit) +{ + return bitmap[bit / 32] & (1 << (bit & 31)); +} + +static void oxygen_restore_ac97(struct oxygen *chip, unsigned int codec) +{ + unsigned int i; + + oxygen_write_ac97(chip, codec, AC97_RESET, 0); + msleep(1); + for (i = 1; i < 0x40; ++i) + if (is_bit_set(ac97_registers_to_restore[codec], i)) + oxygen_write_ac97(chip, codec, i * 2, + chip->saved_ac97_registers[codec][i]); +} + +int oxygen_pci_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + unsigned int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + if (pci_enable_device(pci) < 0) { + snd_printk(KERN_ERR "cannot reenable device"); + snd_card_disconnect(card); + return -EIO; + } + pci_set_master(pci); + + oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); + for (i = 0; i < OXYGEN_IO_SIZE; ++i) + if (is_bit_set(registers_to_restore, i)) + oxygen_write8(chip, i, chip->saved_registers._8[i]); + if (chip->has_ac97_0) + oxygen_restore_ac97(chip, 0); + if (chip->has_ac97_1) + oxygen_restore_ac97(chip, 1); + + if (chip->model->resume) + chip->model->resume(chip); + + oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +EXPORT_SYMBOL(oxygen_pci_resume); +#endif /* CONFIG_PM */ diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index b17c405e069d..9680aa35f81f 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -517,6 +517,7 @@ static int oxygen_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_SUSPEND: pausing = 0; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index abd5313d19b5..9a2c16bf94e0 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -359,6 +359,18 @@ static void xonar_dx_cleanup(struct oxygen *chip) oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); } +static void xonar_d2_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + +static void xonar_dx_resume(struct oxygen *chip) +{ + cs43xx_init(chip); + xonar_enable_output(chip); +} + static void set_pcm1796_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -551,6 +563,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -579,6 +593,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_d2_control_filter, .mixer_init = xonar_mixer_init, .cleanup = xonar_cleanup, + .suspend = xonar_cleanup, + .resume = xonar_d2_resume, .set_dac_params = set_pcm1796_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_pcm1796_volume, @@ -608,6 +624,8 @@ static const struct oxygen_model xonar_models[] = { .control_filter = xonar_dx_control_filter, .mixer_init = xonar_dx_mixer_init, .cleanup = xonar_dx_cleanup, + .suspend = xonar_dx_cleanup, + .resume = xonar_dx_resume, .set_dac_params = set_cs43xx_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, @@ -652,6 +670,10 @@ static struct pci_driver xonar_driver = { .id_table = xonar_ids, .probe = xonar_probe, .remove = __devexit_p(oxygen_pci_remove), +#ifdef CONFIG_PM + .suspend = oxygen_pci_suspend, + .resume = oxygen_pci_resume, +#endif }; static int __init alsa_card_xonar_init(void) -- cgit v1.2.3 From d55d7a1cbbd069f8368ec5c67480d319e7b227b9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:25:39 +0200 Subject: [ALSA] oxygen: add symbols for buffer/period size constraints Introduce symbols for the buffer/period size constraints so that their limits and relationships become clearer. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen_pcm.c | 46 ++++++++++++++++++++++++++++--------------- 1 file changed, 30 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 9680aa35f81f..09a16e459de9 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -24,6 +24,16 @@ #include #include "oxygen.h" +/* most DMA channels have a 16-bit counter for 32-bit words */ +#define BUFFER_BYTES_MAX ((1 << 16) * 4) +/* the multichannel DMA channel has a 24-bit counter */ +#define BUFFER_BYTES_MAX_MULTICH ((1 << 24) * 4) + +#define PERIOD_BYTES_MIN 64 + +#define DEFAULT_BUFFER_BYTES (BUFFER_BYTES_MAX / 2) +#define DEFAULT_BUFFER_BYTES_MULTICH (1024 * 1024) + static const struct snd_pcm_hardware oxygen_stereo_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -44,11 +54,11 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -70,11 +80,11 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .rate_max = 192000, .channels_min = 2, .channels_max = 8, - .buffer_bytes_max = 2048 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 256 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2, .periods_min = 2, - .periods_max = 16384, + .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_ac97_hardware = { .info = SNDRV_PCM_INFO_MMAP | @@ -88,11 +98,11 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { .rate_max = 48000, .channels_min = 2, .channels_max = 2, - .buffer_bytes_max = 256 * 1024, - .period_bytes_min = 128, - .period_bytes_max = 128 * 1024, + .buffer_bytes_max = BUFFER_BYTES_MAX, + .period_bytes_min = PERIOD_BYTES_MIN, + .period_bytes_max = BUFFER_BYTES_MAX / 2, .periods_min = 2, - .periods_max = 2048, + .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware *const oxygen_hardware[PCM_COUNT] = { @@ -664,12 +674,14 @@ int oxygen_pcm_init(struct oxygen *chip) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 512 * 1024, 2048 * 1024); + DEFAULT_BUFFER_BYTES_MULTICH, + BUFFER_BYTES_MAX_MULTICH); if (ins) snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } outs = !!(chip->model->pcm_dev_cfg & PLAYBACK_1_TO_SPDIF); @@ -689,7 +701,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, "Digital"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } if (chip->has_ac97_1) { @@ -719,7 +732,8 @@ int oxygen_pcm_init(struct oxygen *chip) strcpy(pcm->name, outs ? "Front Panel" : "Analog 2"); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), - 128 * 1024, 256 * 1024); + DEFAULT_BUFFER_BYTES, + BUFFER_BYTES_MAX); } return 0; } -- cgit v1.2.3 From ca1f30ad6c3f002d1d9b9355a53b8bbf2fe70588 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 13 May 2008 09:26:01 +0200 Subject: [ALSA] virtuoso: restrict period time to less than 10 s Add a constraint for the period time so that there are less than ten seconds between interrupts so that ALSA does not assume that the device is dead. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/pci/oxygen/oxygen_pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 09a16e459de9..c4ad65a3406f 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -165,6 +165,12 @@ static int oxygen_open(struct snd_pcm_substream *substream, if (err < 0) return err; } + if (channel == PCM_MULTICH) { + err = snd_pcm_hw_constraint_minmax + (runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 0, 8192000); + if (err < 0) + return err; + } snd_pcm_set_sync(substream); chip->streams[channel] = substream; -- cgit v1.2.3 From 02330fbaaded5b603cba112e4bbf62cdadec159a Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Fri, 16 May 2008 12:18:29 +0200 Subject: [ALSA] PCI168 snd-azt3328 Linux driver: another huge update - figured out 'Digital(ly) Enhanced Game Port' functionality, implemented support for it (eliminating gameport polling overhead) - removed optional joystick activation, gameport now enabled unconditionally, since we now support it via the PCI I/O space, not via conflict-prone legacy I/O (which I was thus able to DISABLE now)! - fix playback bug (a muted wave output would get unmuted upon start of playback, of course this is not what we want, thus remember mute state) - implement partial power management: when idle, lower clock rate and disable codec (reduced noise!), and disable gameport circuit when unused - instantiate OPL3 timer, too - much better implementation of snd_azf3328_mixer_write_volume_gradually() - slightly optimized interrupt handling - lots of cleanup This time, I also found a way to verify proper OPL3 operation via MIDI file playback (emulation via synth hardware). Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/azt3328.c | 1192 ++++++++++++++++++++++++++++++++++----------------- sound/pci/azt3328.h | 197 +++++++-- 2 files changed, 968 insertions(+), 421 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5f63af6b88a2..b832333c3023 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005, 2006, 2007 by Andreas Mohr + * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -35,9 +35,20 @@ * (3 weeks' worth of evenings filled with driver work). * (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros) * + * It is quite likely that the AZF3328 chip is the PCI cousin of the + * AZF3318 ("azt1020 pnp", "MM Pro 16") ISA chip, given very similar specs. + * * The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name - * for compatibility reasons) has the following features: + * for compatibility reasons) from Azfin (joint-venture of Aztech and Fincitec, + * Fincitec acquired by National Semiconductor in 2002, together with the + * Fincitec-related company ARSmikro) has the following features: * + * - compatibility & compliance: + * - Microsoft PC 97 ("PC 97 Hardware Design Guide", + * http://www.microsoft.com/whdc/archive/pcguides.mspx) + * - Microsoft PC 98 Baseline Audio + * - MPU401 UART + * - Sound Blaster Emulation (DOS Box) * - builtin AC97 conformant codec (SNR over 80dB) * Note that "conformant" != "compliant"!! this chip's mixer register layout * *differs* from the standard AC97 layout: @@ -48,21 +59,28 @@ * addresses illegally. So far unfortunately it looks like the very flexible * ALSA AC97 support is still not enough to easily compensate for such a * grave layout violation despite all tweaks and quirks mechanisms it offers. - * - builtin genuine OPL3 + * - builtin genuine OPL3 - verified to work fine, 20080506 * - full duplex 16bit playback/record at independent sampling rate - * - MPU401 (+ legacy address support) FIXME: how to enable legacy addr?? + * - MPU401 (+ legacy address support, claimed by one official spec sheet) + * FIXME: how to enable legacy addr?? * - game port (legacy address support) - * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - builtin DirectInput support, helps reduce CPU overhead (interrupt-driven - * features supported) + * features supported). - See common term "Digital Enhanced Game Port"... + * (probably DirectInput 3.0 spec - confirm) + * - builtin 3D enhancement (said to be YAMAHA Ymersion) * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) - * - I2S serial port for external DAC + * - I2S serial output port for external DAC * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) * - supports programmable Sub-vendor and Sub-system ID * required for Microsoft's logo compliance (FIXME: where?) + * At least the Trident 4D Wave DX has one bit somewhere + * to enable writes to PCI subsystem VID registers, that should be it. + * This might easily be in extended PCI reg space, since PCI168 also has + * some custom data starting at 0x80. What kind of config settings + * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms * * Note that this driver now is actually *better* than the Windows driver, @@ -74,6 +92,24 @@ * - "timidity -iAv -B2,8 -Os -EFreverb=0" * - "pmidi -p 128:0 jazz.mid" * + * OPL3 hardware playback testing, try something like: + * cat /proc/asound/hwdep + * and + * aconnect -o + * Then use + * sbiload -Dhw:x,y --opl3 /usr/share/sounds/opl3/std.o3 ......./drums.o3 + * where x,y is the xx-yy number as given in hwdep. + * Then try + * pmidi -p a:b jazz.mid + * where a:b is the client number plus 0 usually, as given by aconnect above. + * Oh, and make sure to unmute the FM mixer control (doh!) + * NOTE: power use during OPL3 playback is _VERY_ high (70W --> 90W!) + * despite no CPU activity, possibly due to hindering ACPI idling somehow. + * Shouldn't be a problem of the AZF3328 chip itself, I'd hope. + * Higher PCM / FM mixer levels seem to conflict (causes crackling), + * at least sometimes. Maybe even use with hardware sequencer timer above :) + * adplay/adplug-utils might soon offer hardware-based OPL3 playback, too. + * * Certain PCI versions of this card are susceptible to DMA traffic underruns * in some systems (resulting in sound crackling/clicking/popping), * probably because they don't have a DMA FIFO buffer or so. @@ -87,6 +123,8 @@ * better than a VIA, yet ironically I still get crackling, like many other * people with the same chipset. * Possible remedies: + * - use speaker (amplifier) output instead of headphone output + * (in case crackling is due to overloaded output clipping) * - plug card into a different PCI slot, preferrably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead @@ -94,18 +132,23 @@ * - fiddle with PCI latency settings (setpci -v -s BUSID latency_timer=XX) * Not too helpful. * - Disable ACPI/power management/"Auto Detect RAM/PCI Clk" in BIOS - * + * * BUGS - * - full-duplex might *still* be problematic, not fully tested recently + * - full-duplex might *still* be problematic, however a recent test was fine * - (non-bug) "Bass/Treble or 3D settings don't work" - they do get evaluated * if you set PCM output switch to "pre 3D" instead of "post 3D". * If this can't be set, then get a mixer application that Isn't Stupid (tm) * (e.g. kmix, gamix) - unfortunately several are!! - * + * - locking is not entirely clean, especially the audio stream activity + * ints --> may be racy + * - an _unconnected_ secondary joystick at the gameport will be reported + * to be "active" (floating values, not precisely -1) due to the way we need + * to read the Digital Enhanced Game Port. Not sure whether it is fixable. + * * TODO * - test MPU401 MIDI playback etc. - * - add some power micro-management (disable various units of the card - * as long as they're unused). However this requires I/O ports which I + * - add more power micro-management (disable various units of the card + * as long as they're unused). However this requires more I/O ports which I * haven't figured out yet and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... @@ -113,6 +156,7 @@ * - figure out some cleverly evil scheme to possibly make ALSA AC97 code * fully accept our quite incompatible ""AC97"" mixer and thus save some * code (but I'm not too optimistic that doing this is possible at all) + * - use MMIO (memory-mapped I/O)? Slightly faster access, e.g. for gameport. */ #include @@ -138,7 +182,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #if defined(CONFIG_GAMEPORT) || (defined(MODULE) && defined(CONFIG_GAMEPORT_MODULE)) -#define SUPPORT_JOYSTICK 1 +#define SUPPORT_GAMEPORT 1 #endif #define DEBUG_MISC 0 @@ -147,13 +191,14 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define DEBUG_PLAY_REC 0 #define DEBUG_IO 0 #define DEBUG_TIMER 0 +#define DEBUG_GAME 0 #define MIXER_TESTING 0 #if DEBUG_MISC #define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgmisc(format, args...) -#endif +#endif #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) @@ -163,25 +208,31 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() #define snd_azf3328_dbgcallleave() -#endif +#endif #if DEBUG_MIXER #define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) -#endif +#endif #if DEBUG_PLAY_REC #define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgplay(format, args...) -#endif +#endif #if DEBUG_MISC #define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) -#endif +#endif + +#if DEBUG_GAME +#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#else +#define snd_azf3328_dbggame(format, args...) +#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ module_param_array(index, int, NULL, 0444); @@ -195,39 +246,44 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable AZF3328 soundcard."); -#ifdef SUPPORT_JOYSTICK -static int joystick[SNDRV_CARDS]; -module_param_array(joystick, bool, NULL, 0444); -MODULE_PARM_DESC(joystick, "Enable joystick for AZF3328 soundcard."); -#endif - static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); +struct snd_azf3328_audio_stream { + struct snd_pcm_substream *substream; + int enabled; + int running; + unsigned long portbase; +}; + +enum snd_azf3328_stream_index { + AZF_PLAYBACK = 0, + AZF_CAPTURE = 1, +}; + struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_port; - unsigned long io2_port; - unsigned long mpu_port; - unsigned long synth_port; - unsigned long mixer_port; + + unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long game_io; /* usually 0xb400, size 8 */ + unsigned long mpu_io; /* usually 0xb800, size 4 */ + unsigned long opl3_io; /* usually 0xbc00, size 8 */ + unsigned long mixer_io; /* usually 0xc000, size 64 */ spinlock_t reg_lock; struct snd_timer *timer; - + struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int is_playing; - unsigned int is_recording; + struct snd_azf3328_audio_stream audio_stream[2]; struct snd_card *card; struct snd_rawmidi *rmidi; -#ifdef SUPPORT_JOYSTICK +#ifdef SUPPORT_GAMEPORT struct gameport *gameport; + int axes[4]; #endif struct pci_dev *pci; @@ -236,10 +292,10 @@ struct snd_azf3328 { #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ - u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2]; - u16 saved_regs_io2 [AZF_IO_SIZE_IO2_PM / 2]; - u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; - u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2]; + u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2]; + u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; + u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; + u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2]; u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; #endif }; @@ -252,126 +308,181 @@ static const struct pci_device_id snd_azf3328_ids[] = { MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); + +static int +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) +{ + u8 prev = inb(reg), new; + + new = (do_set) ? (prev|mask) : (prev & ~mask); + /* we need to always write the new value no matter whether it differs + * or not, since some register bits don't indicate their setting */ + outb(new, reg); + if (new != prev) + return 1; + + return 0; +} + +static int +snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) +{ + u16 prev = inw(reg), new; + + new = (do_set) ? (prev|mask) : (prev & ~mask); + /* we need to always write the new value no matter whether it differs + * or not, since some register bits don't indicate their setting */ + outw(new, reg); + if (new != prev) + return 1; + + return 0; +} + static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outb(value, chip->codec_port + reg); + outb(value, chip->codec_io + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->codec_port + reg); + return inb(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->codec_port + reg); + outw(value, chip->codec_io + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) +{ + return inw(chip->codec_io + reg); +} + +static inline void +snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) +{ + outl(value, chip->codec_io + reg); +} + +static inline u32 +snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->codec_port + reg); + return inl(chip->codec_io + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, int reg, u32 value) +snd_azf3328_game_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { - outl(value, chip->codec_port + reg); + outb(value, chip->game_io + reg); } static inline void -snd_azf3328_io2_outb(const struct snd_azf3328 *chip, int reg, u8 value) +snd_azf3328_game_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outb(value, chip->io2_port + reg); + outw(value, chip->game_io + reg); } static inline u8 -snd_azf3328_io2_inb(const struct snd_azf3328 *chip, int reg) +snd_azf3328_game_inb(const struct snd_azf3328 *chip, unsigned reg) +{ + return inb(chip->game_io + reg); +} + +static inline u16 +snd_azf3328_game_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inb(chip->io2_port + reg); + return inw(chip->game_io + reg); } static inline void -snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, int reg, u16 value) +snd_azf3328_mixer_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) { - outw(value, chip->mixer_port + reg); + outw(value, chip->mixer_io + reg); } static inline u16 -snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, int reg) +snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) { - return inw(chip->mixer_port + reg); + return inw(chip->mixer_io + reg); } -static void -snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, int reg, int do_mute) +#define AZF_MUTE_BIT 0x80 + +static int +snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, + unsigned reg, int do_mute +) { - unsigned long portbase = chip->mixer_port + reg + 1; - unsigned char oldval; + unsigned long portbase = chip->mixer_io + reg + 1; + int updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ - oldval = inb(portbase); - if (do_mute) - oldval |= 0x80; - else - oldval &= ~0x80; - outb(oldval, portbase); + updated = snd_azf3328_io_reg_setb(portbase, AZF_MUTE_BIT, do_mute); + + /* indicate whether it was muted before */ + return (do_mute) ? !updated : updated; } static void -snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg, unsigned char dst_vol_left, unsigned char dst_vol_right, int chan_sel, int delay) +snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, + unsigned reg, + unsigned char dst_vol_left, + unsigned char dst_vol_right, + int chan_sel, int delay +) { - unsigned long portbase = chip->mixer_port + reg; + unsigned long portbase = chip->mixer_io + reg; unsigned char curr_vol_left = 0, curr_vol_right = 0; - int left_done = 0, right_done = 0; - + int left_change = 0, right_change = 0; + snd_azf3328_dbgcallenter(); - if (chan_sel & SET_CHAN_LEFT) + + if (chan_sel & SET_CHAN_LEFT) { curr_vol_left = inb(portbase + 1); - else - left_done = 1; - if (chan_sel & SET_CHAN_RIGHT) + + /* take care of muting flag contained in left channel */ + if (curr_vol_left & AZF_MUTE_BIT) + dst_vol_left |= AZF_MUTE_BIT; + else + dst_vol_left &= ~AZF_MUTE_BIT; + + left_change = (curr_vol_left > dst_vol_left) ? -1 : 1; + } + + if (chan_sel & SET_CHAN_RIGHT) { curr_vol_right = inb(portbase + 0); - else - right_done = 1; - - /* take care of muting flag (0x80) contained in left channel */ - if (curr_vol_left & 0x80) - dst_vol_left |= 0x80; - else - dst_vol_left &= ~0x80; + + right_change = (curr_vol_right > dst_vol_right) ? -1 : 1; + } do { - if (!left_done) { - if (curr_vol_left > dst_vol_left) - curr_vol_left--; - else - if (curr_vol_left < dst_vol_left) - curr_vol_left++; - else - left_done = 1; - outb(curr_vol_left, portbase + 1); + if (left_change) { + if (curr_vol_left != dst_vol_left) { + curr_vol_left += left_change; + outb(curr_vol_left, portbase + 1); + } else + left_change = 0; } - if (!right_done) { - if (curr_vol_right > dst_vol_right) - curr_vol_right--; - else - if (curr_vol_right < dst_vol_right) - curr_vol_right++; - else - right_done = 1; + if (right_change) { + if (curr_vol_right != dst_vol_right) { + curr_vol_right += right_change; + /* during volume change, the right channel is crackling * somewhat more than the left channel, unfortunately. * This seems to be a hardware issue. */ - outb(curr_vol_right, portbase + 0); + outb(curr_vol_right, portbase + 0); + } else + right_change = 0; } if (delay) mdelay(delay); - } while ((!left_done) || (!right_done)); + } while ((left_change) || (right_change)); snd_azf3328_dbgcallleave(); } @@ -379,7 +490,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg * general mixer element */ struct azf3328_mixer_reg { - unsigned int reg; + unsigned reg; unsigned int lchan_shift, rchan_shift; unsigned int mask; unsigned int invert: 1; @@ -544,13 +655,14 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, "Mix", "Mic" }; static const char * const texts3[] = { - "Mic", "CD", "Video", "Aux", + "Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone" }; static const char * const texts4[] = { "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; + const char *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -561,18 +673,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]); + p = texts1[uinfo->value.enumerated.item]; break; case 9: /* mono sel source */ - strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]); + p = texts2[uinfo->value.enumerated.item]; break; case 15: /* PCM Out Path */ - strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]); + p = texts4[uinfo->value.enumerated.item]; break; } } else - strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item] -); + if (reg.reg == IDX_MIXER_REC_SELECT) + p = texts3[uinfo->value.enumerated.item]; + + strcpy(uinfo->value.enumerated.name, p); return 0; } @@ -583,7 +697,7 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned short val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); val = snd_azf3328_mixer_inw(chip, reg.reg); if (reg.reg == IDX_MIXER_REC_SELECT) { @@ -605,7 +719,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; unsigned int oreg, nreg, val; - + snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); val = oreg; @@ -717,15 +831,16 @@ snd_azf3328_mixer_new(struct snd_azf3328 *chip) snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); /* mute and zero volume channels */ - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_init_values); ++idx) { snd_azf3328_mixer_outw(chip, snd_azf3328_init_values[idx][0], snd_azf3328_init_values[idx][1]); } - + /* add mixer controls */ sw = snd_azf3328_mixer_controls; - for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); idx++, sw++) { + for (idx = 0; idx < ARRAY_SIZE(snd_azf3328_mixer_controls); + ++idx, ++sw) { if ((err = snd_ctl_add(chip->card, snd_ctl_new1(sw, chip))) < 0) return err; } @@ -757,8 +872,8 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) } static void -snd_azf3328_setfmt(struct snd_azf3328 *chip, - unsigned int reg, +snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, + unsigned reg, unsigned int bitrate, unsigned int format_width, unsigned int channels @@ -769,24 +884,25 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallenter(); switch (bitrate) { - case 4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; - case 4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; - case 5512: val |= SOUNDFORMAT_FREQ_5510; break; /* the AZF3328 names it "5510" for some strange reason */ - case 6620: val |= SOUNDFORMAT_FREQ_6620; break; - case 8000: val |= SOUNDFORMAT_FREQ_8000; break; - case 9600: val |= SOUNDFORMAT_FREQ_9600; break; - case 11025: val |= SOUNDFORMAT_FREQ_11025; break; - case 13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; - case 16000: val |= SOUNDFORMAT_FREQ_16000; break; - case 22050: val |= SOUNDFORMAT_FREQ_22050; break; - case 32000: val |= SOUNDFORMAT_FREQ_32000; break; - case 44100: val |= SOUNDFORMAT_FREQ_44100; break; - case 48000: val |= SOUNDFORMAT_FREQ_48000; break; - case 66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; + case AZF_FREQ_4000: val |= SOUNDFORMAT_FREQ_SUSPECTED_4000; break; + case AZF_FREQ_4800: val |= SOUNDFORMAT_FREQ_SUSPECTED_4800; break; + case AZF_FREQ_5512: + /* the AZF3328 names it "5510" for some strange reason */ + val |= SOUNDFORMAT_FREQ_5510; break; + case AZF_FREQ_6620: val |= SOUNDFORMAT_FREQ_6620; break; + case AZF_FREQ_8000: val |= SOUNDFORMAT_FREQ_8000; break; + case AZF_FREQ_9600: val |= SOUNDFORMAT_FREQ_9600; break; + case AZF_FREQ_11025: val |= SOUNDFORMAT_FREQ_11025; break; + case AZF_FREQ_13240: val |= SOUNDFORMAT_FREQ_SUSPECTED_13240; break; + case AZF_FREQ_16000: val |= SOUNDFORMAT_FREQ_16000; break; + case AZF_FREQ_22050: val |= SOUNDFORMAT_FREQ_22050; break; + case AZF_FREQ_32000: val |= SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - val |= SOUNDFORMAT_FREQ_44100; - break; + /* fall-through */ + case AZF_FREQ_44100: val |= SOUNDFORMAT_FREQ_44100; break; + case AZF_FREQ_48000: val |= SOUNDFORMAT_FREQ_48000; break; + case AZF_FREQ_66200: val |= SOUNDFORMAT_FREQ_SUSPECTED_66200; break; } /* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */ /* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */ @@ -805,10 +921,10 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, val |= SOUNDFORMAT_FLAG_16BIT; spin_lock_irqsave(&chip->reg_lock, flags); - + /* set bitrate/format */ snd_azf3328_codec_outw(chip, reg, val); - + /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change * (maybe shutting down DAC/ADC?), thus immediately @@ -830,31 +946,81 @@ snd_azf3328_setfmt(struct snd_azf3328 *chip, snd_azf3328_dbgcallleave(); } +static inline void +snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, + unsigned reg +) +{ + /* choose lowest frequency for low power consumption. + * While this will cause louder noise due to rather coarse frequency, + * it should never matter since output should always + * get disabled properly when idle anyway. */ + snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); +} + +static inline void +snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +{ + /* no idea what exactly is being done here, but I strongly assume it's + * PM related */ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_PAUSE_PLAYBACK_BIT8, + !enable + ); +} + +static void +snd_azf3328_codec_activity(struct snd_azf3328 *chip, + enum snd_azf3328_stream_index stream_type, + int enable +) +{ + int need_change = (chip->audio_stream[stream_type].running != enable); + + snd_azf3328_dbgplay( + "codec_activity: type %d, enable %d, need_change %d\n", + stream_type, enable, need_change + ); + if (need_change) { + enum snd_azf3328_stream_index other = + (stream_type == AZF_PLAYBACK) ? + AZF_CAPTURE : AZF_PLAYBACK; + /* small check to prevent shutting down the other party + * in case it's active */ + if ((enable) || !(chip->audio_stream[other].running)) + snd_azf3328_codec_enable(chip, enable); + + /* ...and adjust clock, too + * (reduce noise and power consumption) */ + if (!enable) + snd_azf3328_codec_setfmt_lowpower( + chip, + chip->audio_stream[stream_type].portbase + + IDX_IO_PLAY_SOUNDFORMAT + ); + } + chip->audio_stream[stream_type].running = enable; +} + static void snd_azf3328_setdmaa(struct snd_azf3328 *chip, long unsigned int addr, unsigned int count, unsigned int size, - int do_recording) + enum snd_azf3328_stream_index stream_type +) { - unsigned long flags, portbase; - unsigned int is_running; - snd_azf3328_dbgcallenter(); - if (do_recording) { - /* access capture registers, i.e. skip playback reg section */ - portbase = chip->codec_port + 0x20; - is_running = chip->is_recording; - } else { - /* access the playback register section */ - portbase = chip->codec_port + 0x00; - is_running = chip->is_playing; - } + if (!chip->audio_stream[stream_type].running) { + /* AZF3328 uses a two buffer pointer DMA playback approach */ + + unsigned long flags, portbase, addr_area2; + + /* width 32bit (prevent overflow): */ + unsigned long count_areas, count_tmp; - /* AZF3328 uses a two buffer pointer DMA playback approach */ - if (!is_running) { - unsigned long addr_area2; - unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */ + portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ @@ -884,11 +1050,11 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 0); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); #endif snd_azf3328_dbgcallleave(); return 0; @@ -906,11 +1072,11 @@ snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, 1); + snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); #endif snd_azf3328_dbgcallleave(); return 0; @@ -923,6 +1089,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; unsigned int status1; + int previously_muted; snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); @@ -930,20 +1097,23 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: snd_azf3328_dbgplay("START PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -951,7 +1121,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 0); + AZF_PLAYBACK); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -978,30 +1148,35 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 1; snd_azf3328_dbgplay("STARTED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME PLAYBACK\n"); /* resume playback if we were active */ - if (chip->is_playing) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_PLAYBACK].running) snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP PLAYBACK\n"); - /* mute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); spin_lock(&chip->reg_lock); - /* stop playback */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + /* stop playback */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); @@ -1013,10 +1188,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); spin_unlock(&chip->reg_lock); - + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* now unmute WaveOut */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); - chip->is_playing = 0; + if (!previously_muted) + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1035,7 +1212,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1057,17 +1234,19 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_dbgplay("START CAPTURE\n"); - snd_azf3328_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - + /* FIXME: clear interrupts or what??? */ snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); @@ -1075,7 +1254,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) snd_azf3328_setdmaa(chip, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), snd_pcm_lib_buffer_bytes(substream), - 1); + AZF_CAPTURE); spin_lock(&chip->reg_lock); #ifdef WIN9X @@ -1102,24 +1281,27 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - chip->is_recording = 1; snd_azf3328_dbgplay("STARTED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_RESUME: snd_azf3328_dbgplay("RESUME CAPTURE\n"); /* resume recording if we were active */ - if (chip->is_recording) + spin_lock(&chip->reg_lock); + if (chip->audio_stream[AZF_CAPTURE].running) snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); + spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: snd_azf3328_dbgplay("STOP CAPTURE\n"); spin_lock(&chip->reg_lock); - /* stop recording */ + /* first, remember current value: */ status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); + /* stop recording */ status1 &= ~DMA_RESUME; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); @@ -1129,8 +1311,8 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) status1 &= ~DMA_PLAY_SOMETHING1; snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); spin_unlock(&chip->reg_lock); - - chip->is_recording = 0; + snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + snd_azf3328_dbgplay("STOPPED CAPTURE\n"); break; case SNDRV_PCM_TRIGGER_SUSPEND: @@ -1149,7 +1331,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } - + snd_azf3328_dbgcallleave(); return result; } @@ -1162,11 +1344,11 @@ snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1183,11 +1365,11 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = inl(chip->codec_port+IDX_IO_REC_DMA_START_1); + bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS); + result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; @@ -1196,27 +1378,233 @@ snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) return frmres; } +/******************************************************************/ + +#ifdef SUPPORT_GAMEPORT +static inline void +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_IRQ_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + GAME_HWCFG_LEGACY_ADDRESS_ENABLE, + enable + ); +} + +static inline void +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +{ + snd_azf3328_io_reg_setw( + chip->codec_io+IDX_IO_6AH, + IO_6A_SOMETHING2_GAMEPORT, + !enable + ); +} + +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + /* + * skeleton handler only + * (we do not want axis reading in interrupt handler - too much load!) + */ + snd_azf3328_dbggame("gameport irq\n"); + + /* this should ACK the gameport IRQ properly, hopefully. */ + snd_azf3328_game_inw(chip, IDX_GAME_AXIS_VALUE); +} + +static int +snd_azf3328_gameport_open(struct gameport *gameport, int mode) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int res; + + snd_azf3328_dbggame("gameport_open, mode %d\n", mode); + switch (mode) { + case GAMEPORT_MODE_COOKED: + case GAMEPORT_MODE_RAW: + res = 0; + break; + default: + res = -1; + break; + } + + snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); + + return res; +} + +static void +snd_azf3328_gameport_close(struct gameport *gameport) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + + snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_axis_circuit_enable(chip, 0); +} + +static int +snd_azf3328_gameport_cooked_read(struct gameport *gameport, + int *axes, + int *buttons +) +{ + struct snd_azf3328 *chip = gameport_get_port_data(gameport); + int i; + u8 val; + unsigned long flags; + + snd_assert(chip, return 0); + + spin_lock_irqsave(&chip->reg_lock, flags); + val = snd_azf3328_game_inb(chip, IDX_GAME_LEGACY_COMPATIBLE); + *buttons = (~(val) >> 4) & 0xf; + + /* ok, this one is a bit dirty: cooked_read is being polled by a timer, + * thus we're atomic and cannot actively wait in here + * (which would be useful for us since it probably would be better + * to trigger a measurement in here, then wait a short amount of + * time until it's finished, then read values of _this_ measurement). + * + * Thus we simply resort to reading values if they're available already + * and trigger the next measurement. + */ + + val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); + if (val & GAME_AXES_SAMPLING_READY) { + for (i = 0; i < 4; ++i) { + /* configure the axis to read */ + val = (i << 4) | 0x0f; + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + chip->axes[i] = snd_azf3328_game_inw( + chip, IDX_GAME_AXIS_VALUE + ); + } + } + + /* trigger next axes sampling, to be evaluated the next time we + * enter this function */ + + /* for some very, very strange reason we cannot enable + * Measurement Ready monitoring for all axes here, + * at least not when only one joystick connected */ + val = 0x03; /* we're able to monitor axes 1 and 2 only */ + snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); + + snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + for (i = 0; i < 4; i++) { + axes[i] = chip->axes[i]; + if (axes[i] == 0xffff) + axes[i] = -1; + } + + snd_azf3328_dbggame("cooked_read: axes %d %d %d %d buttons %d\n", + axes[0], axes[1], axes[2], axes[3], *buttons + ); + + return 0; +} + +static int __devinit +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) +{ + struct gameport *gp; + + int io_port = chip->game_io; + + chip->gameport = gp = gameport_allocate_port(); + if (!gp) { + printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); + return -ENOMEM; + } + + gameport_set_name(gp, "AZF3328 Gameport"); + gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); + gameport_set_dev_parent(gp, &chip->pci->dev); + gp->io = io_port; + gameport_set_port_data(gp, chip); + + gp->open = snd_azf3328_gameport_open; + gp->close = snd_azf3328_gameport_close; + gp->fuzz = 16; /* seems ok */ + gp->cooked_read = snd_azf3328_gameport_cooked_read; + + /* DISABLE legacy address: we don't need it! */ + snd_azf3328_gameport_legacy_address_enable(chip, 0); + + snd_azf3328_gameport_axis_circuit_enable(chip, 0); + + gameport_register_port(chip->gameport); + + return 0; +} + +static void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) +{ + if (chip->gameport) { + gameport_unregister_port(chip->gameport); + chip->gameport = NULL; + } + snd_azf3328_gameport_irq_enable(chip, 0); +} +#else +static inline int +snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } +static inline void +snd_azf3328_gameport_free(struct snd_azf3328 *chip) { } +static inline void +snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) +{ + printk(KERN_WARNING "huh, game port IRQ occurred!?\n"); +} +#endif /* SUPPORT_GAMEPORT */ + +/******************************************************************/ + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; u8 status, which; +#if DEBUG_PLAY_REC static unsigned long irq_count; +#endif status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ - if (!(status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_MPU401|IRQ_TIMER))) + if (!(status & + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + )) return IRQ_NONE; /* must be interrupt for another device */ snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count, + irq_count++ /* debug-only */, snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status); - + if (status & IRQ_TIMER) { - /* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */ + /* snd_azf3328_dbgplay("timer %ld\n", + snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE) + & TIMER_VALUE_MASK + ); */ if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ @@ -1232,11 +1620,16 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->playback_substream) { - snd_pcm_period_elapsed(chip->playback_substream); + if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_PLAYBACK].substream + ); snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_PLAY_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) @@ -1249,16 +1642,23 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->capture_substream) { - snd_pcm_period_elapsed(chip->capture_substream); + if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { + snd_pcm_period_elapsed( + chip->audio_stream[AZF_CAPTURE].substream + ); snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", which, - inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS)); + snd_azf3328_codec_inl( + chip, IDX_IO_REC_DMA_CURRPOS + ) + ); } else snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); } + if (status & IRQ_GAMEPORT) + snd_azf3328_gameport_interrupt(chip); /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { @@ -1268,7 +1668,6 @@ snd_azf3328_interrupt(int irq, void *dev_id) * If so, then I don't know how... */ snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); } - irq_count++; return IRQ_HANDLED; } @@ -1287,8 +1686,8 @@ static const struct snd_pcm_hardware snd_azf3328_playback = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1315,8 +1714,8 @@ static const struct snd_pcm_hardware snd_azf3328_capture = .rates = SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, - .rate_min = 4000, - .rate_max = 66200, + .rate_min = AZF_FREQ_4000, + .rate_max = AZF_FREQ_66200, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 65536, @@ -1329,10 +1728,24 @@ static const struct snd_pcm_hardware snd_azf3328_capture = static unsigned int snd_azf3328_fixed_rates[] = { - 4000, 4800, 5512, 6620, 8000, 9600, 11025, 13240, 16000, 22050, 32000, - 44100, 48000, 66200 }; + AZF_FREQ_4000, + AZF_FREQ_4800, + AZF_FREQ_5512, + AZF_FREQ_6620, + AZF_FREQ_8000, + AZF_FREQ_9600, + AZF_FREQ_11025, + AZF_FREQ_13240, + AZF_FREQ_16000, + AZF_FREQ_22050, + AZF_FREQ_32000, + AZF_FREQ_44100, + AZF_FREQ_48000, + AZF_FREQ_66200 +}; + static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { - .count = ARRAY_SIZE(snd_azf3328_fixed_rates), + .count = ARRAY_SIZE(snd_azf3328_fixed_rates), .list = snd_azf3328_fixed_rates, .mask = 0, }; @@ -1346,7 +1759,7 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->playback_substream = substream; + chip->audio_stream[AZF_PLAYBACK].substream = substream; runtime->hw = snd_azf3328_playback; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1361,7 +1774,7 @@ snd_azf3328_capture_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->capture_substream = substream; + chip->audio_stream[AZF_CAPTURE].substream = substream; runtime->hw = snd_azf3328_capture; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); @@ -1375,7 +1788,7 @@ snd_azf3328_playback_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->playback_substream = NULL; + chip->audio_stream[AZF_PLAYBACK].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1386,7 +1799,7 @@ snd_azf3328_capture_close(struct snd_pcm_substream *substream) struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->capture_substream = NULL; + chip->audio_stream[AZF_CAPTURE].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } @@ -1441,102 +1854,8 @@ snd_azf3328_pcm(struct snd_azf3328 *chip, int device) /******************************************************************/ -#ifdef SUPPORT_JOYSTICK -static int __devinit -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) -{ - struct gameport *gp; - struct resource *r; - - if (!joystick[dev]) - return -ENODEV; - - if (!(r = request_region(0x200, 8, "AZF3328 gameport"))) { - printk(KERN_WARNING "azt3328: cannot reserve joystick ports\n"); - return -EBUSY; - } - - chip->gameport = gp = gameport_allocate_port(); - if (!gp) { - printk(KERN_ERR "azt3328: cannot allocate memory for gameport\n"); - release_and_free_resource(r); - return -ENOMEM; - } - - gameport_set_name(gp, "AZF3328 Gameport"); - gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); - gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = 0x200; - gameport_set_port_data(gp, r); - - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) | LEGACY_JOY); - - gameport_register_port(chip->gameport); - - return 0; -} - -static void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) -{ - if (chip->gameport) { - struct resource *r = gameport_get_port_data(chip->gameport); - - gameport_unregister_port(chip->gameport); - chip->gameport = NULL; - /* disable gameport */ - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); - release_and_free_resource(r); - } -} -#else -static inline int -snd_azf3328_config_joystick(struct snd_azf3328 *chip, int dev) { return -ENOSYS; } -static inline void -snd_azf3328_free_joystick(struct snd_azf3328 *chip) { } -#endif - -/******************************************************************/ - -static int -snd_azf3328_free(struct snd_azf3328 *chip) -{ - if (chip->irq < 0) - goto __end_hw; - - /* reset (close) mixer */ - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); /* first mute master volume */ - snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); - - /* interrupt setup - mask everything (FIXME!) */ - /* well, at least we know how to disable the timer IRQ */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); - - if (chip->irq >= 0) - synchronize_irq(chip->irq); -__end_hw: - snd_azf3328_free_joystick(chip); - if (chip->irq >= 0) - free_irq(chip->irq, chip); - pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - - kfree(chip); - return 0; -} - -static int -snd_azf3328_dev_free(struct snd_device *device) -{ - struct snd_azf3328 *chip = device->device_data; - return snd_azf3328_free(chip); -} - -/******************************************************************/ - -/*** NOTE: the physical timer resolution actually is 1024000 ticks per second, +/*** NOTE: the physical timer resolution actually is 1024000 ticks per second + *** (probably derived from main crystal via a divider of 24), *** but announcing those attributes to user-space would make programs *** configure the timer to a 1 tick value, resulting in an absolutely fatal *** timer IRQ storm. @@ -1564,7 +1883,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) delay = 49; /* minimum time is 49 ticks */ } snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); - delay |= TIMER_ENABLE_COUNTDOWN | TIMER_ENABLE_IRQ; + delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1582,7 +1901,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer) chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ - /* FIXME: should we write TIMER_ACK_IRQ here? */ + /* FIXME: should we write TIMER_IRQ_ACK here? */ snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); @@ -1626,9 +1945,10 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) snd_azf3328_timer_hw.resolution *= seqtimer_scaling; snd_azf3328_timer_hw.ticks /= seqtimer_scaling; - if ((err = snd_timer_new(chip->card, "AZF3328", &tid, &timer)) < 0) { + + err = snd_timer_new(chip->card, "AZF3328", &tid, &timer); + if (err < 0) goto out; - } strcpy(timer->name, "AZF3328 timer"); timer->private_data = chip; @@ -1636,6 +1956,8 @@ snd_azf3328_timer(struct snd_azf3328 *chip, int device) chip->timer = timer; + snd_azf3328_timer_stop(timer); + err = 0; out: @@ -1645,10 +1967,44 @@ out: /******************************************************************/ +static int +snd_azf3328_free(struct snd_azf3328 *chip) +{ + if (chip->irq < 0) + goto __end_hw; + + /* reset (close) mixer: + * first mute master volume, then reset + */ + snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); + snd_azf3328_mixer_outw(chip, IDX_MIXER_RESET, 0x0000); + + snd_azf3328_timer_stop(chip->timer); + snd_azf3328_gameport_free(chip); + + if (chip->irq >= 0) + synchronize_irq(chip->irq); +__end_hw: + if (chip->irq >= 0) + free_irq(chip->irq, chip); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + + kfree(chip); + return 0; +} + +static int +snd_azf3328_dev_free(struct snd_device *device) +{ + struct snd_azf3328 *chip = device->device_data; + return snd_azf3328_free(chip); +} + #if 0 /* check whether a bit can be modified */ static void -snd_azf3328_test_bit(unsigned int reg, int bit) +snd_azf3328_test_bit(unsigned unsigned reg, int bit) { unsigned char val, valoff, valon; @@ -1659,42 +2015,74 @@ snd_azf3328_test_bit(unsigned int reg, int bit) outb(val|(1 << bit), reg); valon = inb(reg); - + outb(val, reg); - printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon); + printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", + reg, bit, val, valoff, valon + ); } #endif -#if DEBUG_MISC -static void +static inline void snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) { +#if DEBUG_MISC u16 tmp; - snd_azf3328_dbgmisc("codec_port 0x%lx, io2_port 0x%lx, mpu_port 0x%lx, synth_port 0x%lx, mixer_port 0x%lx, irq %d\n", chip->codec_port, chip->io2_port, chip->mpu_port, chip->synth_port, chip->mixer_port, chip->irq); - - snd_azf3328_dbgmisc("io2 %02x %02x %02x %02x %02x %02x\n", snd_azf3328_io2_inb(chip, 0), snd_azf3328_io2_inb(chip, 1), snd_azf3328_io2_inb(chip, 2), snd_azf3328_io2_inb(chip, 3), snd_azf3328_io2_inb(chip, 4), snd_azf3328_io2_inb(chip, 5)); - - for (tmp=0; tmp <= 0x01; tmp += 1) - snd_azf3328_dbgmisc("0x%02x: opl 0x%04x, mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, mpu330 0x%04x\n", tmp, inb(0x388 + tmp), inb(0x300 + tmp), inb(0x310 + tmp), inb(0x320 + tmp), inb(0x330 + tmp)); + snd_azf3328_dbgmisc( + "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", + chip->codec_io, chip->game_io, chip->mpu_io, + chip->opl3_io, chip->mixer_io, chip->irq + ); + + snd_azf3328_dbgmisc("game %02x %02x %02x %02x %02x %02x\n", + snd_azf3328_game_inb(chip, 0), + snd_azf3328_game_inb(chip, 1), + snd_azf3328_game_inb(chip, 2), + snd_azf3328_game_inb(chip, 3), + snd_azf3328_game_inb(chip, 4), + snd_azf3328_game_inb(chip, 5) + ); + + for (tmp = 0; tmp < 0x07; tmp += 1) + snd_azf3328_dbgmisc("mpu_io 0x%04x\n", inb(chip->mpu_io + tmp)); + + for (tmp = 0; tmp <= 0x07; tmp += 1) + snd_azf3328_dbgmisc("0x%02x: game200 0x%04x, game208 0x%04x\n", + tmp, inb(0x200 + tmp), inb(0x208 + tmp)); + + for (tmp = 0; tmp <= 0x01; tmp += 1) + snd_azf3328_dbgmisc( + "0x%02x: mpu300 0x%04x, mpu310 0x%04x, mpu320 0x%04x, " + "mpu330 0x%04x opl388 0x%04x opl38c 0x%04x\n", + tmp, + inb(0x300 + tmp), + inb(0x310 + tmp), + inb(0x320 + tmp), + inb(0x330 + tmp), + inb(0x388 + tmp), + inb(0x38c + tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", tmp, snd_azf3328_codec_inw(chip, tmp)); + snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", + tmp, snd_azf3328_codec_inw(chip, tmp) + ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) - snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", tmp, snd_azf3328_mixer_inw(chip, tmp)); + snd_azf3328_dbgmisc("mixer 0x%02x: 0x%04x\n", + tmp, snd_azf3328_mixer_inw(chip, tmp) + ); +#endif /* DEBUG_MISC */ } -#else -static inline void -snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) {} -#endif static int __devinit snd_azf3328_create(struct snd_card *card, - struct pci_dev *pci, - unsigned long device_type, - struct snd_azf3328 ** rchip) + struct pci_dev *pci, + unsigned long device_type, + struct snd_azf3328 **rchip) { struct snd_azf3328 *chip; int err; @@ -1705,7 +2093,8 @@ snd_azf3328_create(struct snd_card *card, *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; chip = kzalloc(sizeof(*chip), GFP_KERNEL); @@ -1721,20 +2110,25 @@ snd_azf3328_create(struct snd_card *card, /* check if we can restrict PCI DMA transfers to 24 bits */ if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "24bit PCI busmaster DMA\n" + ); err = -ENXIO; goto out_err; } - if ((err = pci_request_regions(pci, "Aztech AZF3328")) < 0) { + err = pci_request_regions(pci, "Aztech AZF3328"); + if (err < 0) goto out_err; - } - chip->codec_port = pci_resource_start(pci, 0); - chip->io2_port = pci_resource_start(pci, 1); - chip->mpu_port = pci_resource_start(pci, 2); - chip->synth_port = pci_resource_start(pci, 3); - chip->mixer_port = pci_resource_start(pci, 4); + chip->codec_io = pci_resource_start(pci, 0); + chip->game_io = pci_resource_start(pci, 1); + chip->mpu_io = pci_resource_start(pci, 2); + chip->opl3_io = pci_resource_start(pci, 3); + chip->mixer_io = pci_resource_start(pci, 4); + + chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; + chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -1747,29 +2141,29 @@ snd_azf3328_create(struct snd_card *card, synchronize_irq(chip->irq); snd_azf3328_debug_show_ports(chip); - - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) goto out_err; - } /* create mixer interface & switches */ - if ((err = snd_azf3328_mixer_new(chip)) < 0) + err = snd_azf3328_mixer_new(chip); + if (err < 0) goto out_err; -#if 0 - /* set very low bitrate to reduce noise and power consumption? */ - snd_azf3328_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, 5512, 8, 1); -#endif + /* shutdown codecs to save power */ + /* have snd_azf3328_codec_activity() act properly */ + chip->audio_stream[AZF_PLAYBACK].running = 1; + snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); /* standard chip init stuff */ - /* default IRQ init value */ + /* default IRQ init value */ tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; spin_lock_irq(&chip->reg_lock); snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x00); /* disable timer */ spin_unlock_irq(&chip->reg_lock); snd_card_set_dev(card, &pci->dev); @@ -1805,52 +2199,61 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0 ); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); - if ((err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip)) < 0) { + err = snd_azf3328_create(card, pci, pci_id->driver_data, &chip); + if (err < 0) goto out_err; - } card->private_data = chip; - if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401, - chip->mpu_port, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { - snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port); + err = snd_mpu401_uart_new( + card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED, + pci->irq, 0, &chip->rmidi + ); + if (err < 0) { + snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", + chip->mpu_io + ); goto out_err; } - if ((err = snd_azf3328_timer(chip, 0)) < 0) { + err = snd_azf3328_timer(chip, 0); + if (err < 0) goto out_err; - } - if ((err = snd_azf3328_pcm(chip, 0)) < 0) { + err = snd_azf3328_pcm(chip, 0); + if (err < 0) goto out_err; - } - if (snd_opl3_create(card, chip->synth_port, chip->synth_port+2, + if (snd_opl3_create(card, chip->opl3_io, chip->opl3_io+2, OPL3_HW_AUTO, 1, &opl3) < 0) { snd_printk(KERN_ERR "azf3328: no OPL3 device at 0x%lx-0x%lx?\n", - chip->synth_port, chip->synth_port+2 ); + chip->opl3_io, chip->opl3_io+2 + ); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + /* need to use IDs 1, 2 since ID 0 is snd_azf3328_timer above */ + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) goto out_err; - } } opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_port, chip->irq); + card->shortname, chip->codec_io, chip->irq); - if ((err = snd_card_register(card)) < 0) { + err = snd_card_register(card); + if (err < 0) goto out_err; - } #ifdef MODULE printk( @@ -1861,19 +2264,18 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) 1024000 / seqtimer_scaling, seqtimer_scaling); #endif - if (snd_azf3328_config_joystick(chip, dev) < 0) - snd_azf3328_io2_outb(chip, IDX_IO2_LEGACY_ADDR, - snd_azf3328_io2_inb(chip, IDX_IO2_LEGACY_ADDR) & ~LEGACY_JOY); + snd_azf3328_gameport(chip, dev); pci_set_drvdata(pci, card); dev++; err = 0; goto out; - + out_err: + snd_printk(KERN_ERR "azf3328: something failed, exiting\n"); snd_card_free(card); - + out: snd_azf3328_dbgcallleave(); return err; @@ -1894,27 +2296,27 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - + snd_pcm_suspend_all(chip->pcm); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); /* make sure to disable master volume etc. to prevent looping sound */ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2); + + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2); pci_disable_device(pci); pci_save_state(pci); @@ -1927,7 +2329,7 @@ snd_azf3328_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - int reg; + unsigned reg; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -1939,23 +2341,21 @@ snd_azf3328_resume(struct pci_dev *pci) } pci_set_master(pci); - for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++) - outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++) - outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++) - outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++) - outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++) - outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) + outw(chip->saved_regs_game[reg], chip->game_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) + outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) + outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) + outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); + for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) + outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -#endif - - +#endif /* CONFIG_PM */ static struct pci_driver driver = { diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 679fa992e2bc..3448fd626f80 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -54,7 +54,10 @@ #define SOUNDFORMAT_XTAL1 0x00 #define SOUNDFORMAT_XTAL2 0x01 /* all _SUSPECTED_ values are not used by Windows drivers, so we don't - * have any hard facts, only rough measurements */ + * have any hard facts, only rough measurements. + * All we know is that the crystal used on the board has 24.576MHz, + * like many soundcards (which results in the frequencies below when + * using certain divider values selected by the values below) */ #define SOUNDFORMAT_FREQ_SUSPECTED_4000 0x0c | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_SUSPECTED_4800 0x0a | SOUNDFORMAT_XTAL1 #define SOUNDFORMAT_FREQ_5510 0x0c | SOUNDFORMAT_XTAL2 @@ -72,6 +75,26 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 +/* define frequency helpers, for maximum value safety */ +enum { +#define AZF_FREQ(rate) AZF_FREQ_##rate = rate + AZF_FREQ(4000), + AZF_FREQ(4800), + AZF_FREQ(5512), + AZF_FREQ(6620), + AZF_FREQ(8000), + AZF_FREQ(9600), + AZF_FREQ(11025), + AZF_FREQ(13240), + AZF_FREQ(16000), + AZF_FREQ(22050), + AZF_FREQ(32000), + AZF_FREQ(44100), + AZF_FREQ(48000), + AZF_FREQ(66200), +#undef AZF_FREQ +} AZF_FREQUENCIES; + /** recording area (see also: playback bit flag definitions) **/ #define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ #define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ @@ -97,40 +120,164 @@ /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ - #define TIMER_VALUE_MASK 0x000fffffUL /* timer countdown value; triggers IRQ when timer is finished */ - #define TIMER_ENABLE_COUNTDOWN 0x01000000UL /* activate the timer countdown */ - #define TIMER_ENABLE_IRQ 0x02000000UL /* trigger timer IRQ on zero transition */ - #define TIMER_ACK_IRQ 0x04000000UL /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) had 0x0020 set upon IRQ handler */ + /* timer countdown value; triggers IRQ when timer is finished */ + #define TIMER_VALUE_MASK 0x000fffffUL + /* activate timer countdown */ + #define TIMER_COUNTDOWN_ENABLE 0x01000000UL + /* trigger timer IRQ on zero transition */ + #define TIMER_IRQ_ENABLE 0x02000000UL + /* being set in IRQ handler in case port 0x00 (hmm, not port 0x64!?!?) + * had 0x0020 set upon IRQ handler */ + #define TIMER_IRQ_ACK 0x04000000UL #define IDX_IO_IRQSTATUS 0x64 - #define IRQ_PLAYBACK 0x0001 - #define IRQ_RECORDING 0x0002 - #define IRQ_MPU401 0x0010 - #define IRQ_TIMER 0x0020 /* DirectX timer */ - #define IRQ_UNKNOWN1 0x0040 /* probably unused, or possibly I2S port? or gameport IRQ? */ - #define IRQ_UNKNOWN2 0x0080 /* probably unused, or possibly I2S port? or gameport IRQ? */ + /* some IRQ bit in here might also be used to signal a power-management timer + * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing). + * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which + * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */ + + #define IRQ_PLAYBACK 0x0001 + #define IRQ_RECORDING 0x0002 + #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ + #define IRQ_MPU401 0x0010 + #define IRQ_TIMER 0x0020 /* DirectX timer */ + #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */ + #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ -#define IDX_IO_SOME_VALUE 0x68 /* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */ -#define IDX_IO_6AH 0x6A /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */ - #define IO_6A_PAUSE_PLAYBACK 0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */ -#define IDX_IO_6CH 0x6C -#define IDX_IO_6EH 0x6E /* writing 0xffff returns 0x83fe */ -/* further I/O indices not saved/restored, so probably not used */ + /* this is set to e.g. 0x3ff or 0x300, and writable; + * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */ +#define IDX_IO_SOME_VALUE 0x68 + #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ + #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ + /* umm, nope, behaviour of these bits changes depending on what we wrote + * to 0x6b!! */ + +/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); + * actually inhibits PCM playback!!! maybe power management??: */ +#define IDX_IO_6AH 0x6A + /* bit 5: enabling this will activate permanent counting of bytes 2/3 + * at gameport I/O (0xb402/3) (equal values each) and cause + * gameport legacy I/O at 0x0200 to be _DISABLED_! + * Is this Digital Enhanced Game Port Enable??? Or maybe it's Testmode + * for Enhanced Digital Gameport (see 4D Wave DX card): */ + #define IO_6A_SOMETHING1_GAMEPORT 0x0020 + /* bit 8; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT8 0x0100 + /* bit 9; sure, this _pauses_ playback (later resumes at same spot!), + * but what the heck is this really about??: */ + #define IO_6A_PAUSE_PLAYBACK_BIT9 0x0200 + /* BIT8 and BIT9 are _NOT_ able to affect OPL3 MIDI playback, + * thus it suggests influence on PCM only!! + * However OTOH there seems to be no bit anywhere around here + * which is able to disable OPL3... */ + /* bit 10: enabling this actually changes values at legacy gameport + * I/O address (0x200); is this enabling of the Digital Enhanced Game Port??? + * Or maybe this simply switches off the NE558 circuit, since enabling this + * still lets us evaluate button states, but not axis states */ + #define IO_6A_SOMETHING2_GAMEPORT 0x0400 + /* writing 0x0300: causes quite some crackling during + * PC activity such as switching windows (PCI traffic?? + * --> FIFO/timing settings???) */ + /* writing 0x0100 plus/or 0x0200 inhibits playback */ + /* since the Windows .INF file has Flag_Enable_JoyStick and + * Flag_Enable_SB_DOS_Emulation directly together, it stands to reason + * that some other bit in this same register might be responsible + * for SB DOS Emulation activation (note that the file did NOT define + * a switch for OPL3!) */ +#define IDX_IO_6CH 0x6C /* unknown; fully read-writable */ +#define IDX_IO_6EH 0x6E + /* writing 0xffff returns 0x83fe (or 0x03fe only). + * writing 0x83 (and only 0x83!!) to 0x6f will cause 0x6c to switch + * from 0000 to ffff. */ +/* further I/O indices not saved/restored and not readable after writing, + * so probably not used */ -/*** I/O 2 area port indices ***/ + +/*** Gameport area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_IO2 0x08 -#define AZF_IO_SIZE_IO2_PM 0x06 +#define AZF_IO_SIZE_GAME 0x08 +#define AZF_IO_SIZE_GAME_PM 0x06 + +enum { + AZF_GAME_LEGACY_IO_PORT = 0x200 +} AZF_GAME_CONFIGS; + +#define IDX_GAME_LEGACY_COMPATIBLE 0x00 + /* in some operation mode, writing anything to this port + * triggers an interrupt: + * yup, that's in case IDX_GAME_01H has one of the + * axis measurement bits enabled + * (and of course one needs to have GAME_HWCFG_IRQ_ENABLE, too) */ + +#define IDX_GAME_AXES_CONFIG 0x01 + /* NOTE: layout of this register awfully similar (read: "identical??") + * to AD1815JS.pdf (p.29) */ + + /* enables axis 1 (X axis) measurement: */ + #define GAME_AXES_ENABLE_1 0x01 + /* enables axis 2 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_2 0x02 + /* enables axis 3 (X axis) measurement: */ + #define GAME_AXES_ENABLE_3 0x04 + /* enables axis 4 (Y axis) measurement: */ + #define GAME_AXES_ENABLE_4 0x08 + /* selects the current axis to read the measured value of + * (at IDX_GAME_AXIS_VALUE): + * 00 = axis 1, 01 = axis 2, 10 = axis 3, 11 = axis 4: */ + #define GAME_AXES_READ_MASK 0x30 + /* enable to have the latch continuously accept ADC values + * (and continuously cause interrupts in case interrupts are enabled); + * AD1815JS.pdf says it's ~16ms interval there: */ + #define GAME_AXES_LATCH_ENABLE 0x40 + /* joystick data (measured axes) ready for reading: */ + #define GAME_AXES_SAMPLING_READY 0x80 + + /* NOTE: other card specs (SiS960 and others!) state that the + * game position latches should be frozen when reading and be freed + * (== reset?) after reading!!! + * Freezing most likely means disabling 0x40 (GAME_AXES_LATCH_ENABLE), + * but how to free the value? */ + /* An internet search for "gameport latch ADC" should provide some insight + * into how to program such a gameport system. */ + + /* writing 0xf0 to 01H once reset both counters to 0, in some special mode!? + * yup, in case 6AH 0x20 is not enabled + * (and 0x40 is sufficient, 0xf0 is not needed) */ + +#define IDX_GAME_AXIS_VALUE 0x02 + /* R: value of currently configured axis (word value!); + * W: trigger axis measurement */ + +#define IDX_GAME_HWCONFIG 0x04 + /* note: bits 4 to 7 are never set (== 0) when reading! + * --> reserved bits? */ + /* enables IRQ notification upon axes measurement ready: */ + #define GAME_HWCFG_IRQ_ENABLE 0x01 + /* these bits choose a different frequency for the + * internal ADC counter increment. + * hmm, seems to be a combo of bits: + * 00 --> standard frequency + * 10 --> 1/2 + * 01 --> 1/20 + * 11 --> 1/200: */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 -#define IDX_IO2_LEGACY_ADDR 0x04 - #define LEGACY_SOMETHING 0x01 /* OPL3?? */ - #define LEGACY_JOY 0x08 + /* enable gameport legacy I/O address (0x200) + * I was unable to locate any configurability for a different address: */ + #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 +/*** MPU401 ***/ #define AZF_IO_SIZE_MPU 0x04 #define AZF_IO_SIZE_MPU_PM 0x04 -#define AZF_IO_SIZE_SYNTH 0x08 -#define AZF_IO_SIZE_SYNTH_PM 0x06 +/*** OPL3 synth ***/ +#define AZF_IO_SIZE_OPL3 0x08 +#define AZF_IO_SIZE_OPL3_PM 0x06 +/* hmm, given that a standard OPL3 has 4 registers only, + * there might be some enhanced functionality lurking at the end + * (especially since register 0x04 has a "non-empty" value 0xfe) */ /*** mixer I/O area port indices ***/ /* (only 0x22 of 0x40 bytes saved/restored by Windows driver) -- cgit v1.2.3 From 9ad593f6d326e7a4664e3856520f6c042f82a37f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 May 2008 12:34:47 +0200 Subject: [ALSA] hda - Fix DMA position inaccuracy Many HD-audio controllers seem inaccurate about the IRQ timing of PCM period updates. This has caused problems on audio quality; e.g. JACK doesn't work with two periods. This patch fixes the problem by checking the current DMA position at IRQ handler and delays the period-update via a workq if it's inaccurate. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 124 +++++++++++++++++++++++++++++++++++++++------- 1 file changed, 106 insertions(+), 18 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b3a618eb42cd..6ba7ac01d9f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -285,6 +285,7 @@ struct azx_dev { u32 *posbuf; /* position buffer pointer */ unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ @@ -301,11 +302,10 @@ struct azx_dev { */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ - /* for sanity check of position buffer */ - unsigned int period_intr; unsigned int opened :1; unsigned int running :1; + unsigned int irq_pending: 1; }; /* CORB/RIRB */ @@ -369,6 +369,9 @@ struct azx { /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ + + /* for pending irqs */ + struct work_struct irq_pending_work; }; /* driver types */ @@ -908,6 +911,8 @@ static void azx_init_pci(struct azx *chip) } +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev); + /* * interrupt handler */ @@ -930,11 +935,18 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_dev = &chip->azx_dev[i]; if (status & azx_dev->sd_int_sta_mask) { azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); - if (azx_dev->substream && azx_dev->running) { - azx_dev->period_intr++; + if (!azx_dev->substream || !azx_dev->running) + continue; + /* check whether this IRQ is really acceptable */ + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); + } else { + /* bogus IRQ, process it later */ + azx_dev->irq_pending = 1; + schedule_work(&chip->irq_pending_work); } } } @@ -973,6 +985,7 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, azx_sd_writel(azx_dev, SD_BDLPU, 0); period_bytes = snd_pcm_lib_period_bytes(substream); + azx_dev->period_bytes = period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1421,27 +1434,16 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +static unsigned int azx_get_position(struct azx *chip, + struct azx_dev *azx_dev) { - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; if (chip->position_fix == POS_FIX_POSBUF || chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = le32_to_cpu(*azx_dev->posbuf); - if (chip->position_fix == POS_FIX_AUTO && - azx_dev->period_intr == 1 && !pos) { - printk(KERN_WARNING - "hda-intel: Invalid position buffer, " - "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_NONE; - goto read_lpib; - } } else { - read_lpib: /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); if (chip->position_fix == POS_FIX_FIFO) @@ -1449,7 +1451,90 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) } if (pos >= azx_dev->bufsize) pos = 0; - return bytes_to_frames(substream->runtime, pos); + return pos; +} + +static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + struct azx_dev *azx_dev = get_azx_dev(substream); + return bytes_to_frames(substream->runtime, + azx_get_position(chip, azx_dev)); +} + +/* + * Check whether the current DMA position is acceptable for updating + * periods. Returns non-zero if it's OK. + * + * Many HD-audio controllers appear pretty inaccurate about + * the update-IRQ timing. The IRQ is issued before actually the + * data is processed. So, we need to process it afterwords in a + * workqueue. + */ +static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) +{ + unsigned int pos; + + pos = azx_get_position(chip, azx_dev); + if (chip->position_fix == POS_FIX_AUTO) { + if (!pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->position_fix = POS_FIX_NONE; + pos = azx_get_position(chip, azx_dev); + } else + chip->position_fix = POS_FIX_POSBUF; + } + + if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) + return 0; /* NG - it's below the period boundary */ + return 1; /* OK, it's fine */ +} + +/* + * The work for pending PCM period updates. + */ +static void azx_irq_pending_work(struct work_struct *work) +{ + struct azx *chip = container_of(work, struct azx, irq_pending_work); + int i, pending; + + for (;;) { + pending = 0; + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) { + struct azx_dev *azx_dev = &chip->azx_dev[i]; + if (!azx_dev->irq_pending || + !azx_dev->substream || + !azx_dev->running) + continue; + if (azx_position_ok(chip, azx_dev)) { + azx_dev->irq_pending = 0; + spin_unlock(&chip->reg_lock); + snd_pcm_period_elapsed(azx_dev->substream); + spin_lock(&chip->reg_lock); + } else + pending++; + } + spin_unlock_irq(&chip->reg_lock); + if (!pending) + return; + cond_resched(); + } +} + +/* clear irq_pending flags and assure no on-going workq */ +static void azx_clear_irq_pending(struct azx *chip) +{ + int i; + + spin_lock_irq(&chip->reg_lock); + for (i = 0; i < chip->num_streams; i++) + chip->azx_dev[i].irq_pending = 0; + spin_unlock_irq(&chip->reg_lock); + flush_scheduled_work(); } static struct snd_pcm_ops azx_pcm_ops = { @@ -1676,6 +1761,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) int i; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); if (chip->initialized) @@ -1732,6 +1818,7 @@ static int azx_free(struct azx *chip) int i; if (chip->initialized) { + azx_clear_irq_pending(chip); for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); azx_stop_chip(chip); @@ -1857,6 +1944,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; chip->msi = enable_msi; + INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); chip->position_fix = check_position_fix(chip, position_fix[dev]); check_probe_mask(chip, dev); -- cgit v1.2.3 From aea3bfbcfb0453217c8da6cfdc1b2394d214bee5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 20 May 2008 14:22:44 +0200 Subject: [ALSA] ice1724: fix MIDI The VT1724 MIDI port is not MPU-401 compatible; remove the hacks that try to make the MPU-401 library work with it, and just use some simple device-specific code. Signed-off-by: Clemens Ladisch Tested-by: Pavel Hofman --- sound/pci/Kconfig | 2 +- sound/pci/ice1712/envy24ht.h | 10 +- sound/pci/ice1712/ice1712.h | 2 + sound/pci/ice1712/ice1724.c | 213 +++++++++++++++++++++++++++++++++---------- 4 files changed, 177 insertions(+), 50 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7e4742109572..d95fbb2b5b9f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -692,7 +692,7 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" depends on SND - select SND_MPU401_UART + select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER help diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index 43b9e3e858be..a0c5e009bb4a 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -93,9 +93,13 @@ enum { #define VT1724_REG_MPU_TXFIFO 0x0a /*byte ro. number of bytes in TX fifo*/ #define VT1724_REG_MPU_RXFIFO 0x0b /*byte ro. number of bytes in RX fifo*/ -//are these 2 the wrong way around? they don't seem to be used yet anyway -#define VT1724_REG_MPU_CTRL 0x0c /* byte */ -#define VT1724_REG_MPU_DATA 0x0d /* byte */ +#define VT1724_REG_MPU_DATA 0x0c /* byte */ +#define VT1724_REG_MPU_CTRL 0x0d /* byte */ +#define VT1724_MPU_UART 0x01 +#define VT1724_MPU_TX_EMPTY 0x02 +#define VT1724_MPU_TX_FULL 0x04 +#define VT1724_MPU_RX_EMPTY 0x08 +#define VT1724_MPU_RX_FULL 0x10 #define VT1724_REG_MPU_FIFO_WM 0x0e /*byte set the high/low watermarks for RX/TX fifos*/ #define VT1724_MPU_RX_FIFO 0x20 //1=rx fifo watermark 0=tx fifo watermark diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index 3208901c740e..762fbd7a7507 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -333,6 +333,8 @@ struct snd_ice1712 { unsigned int has_spdif: 1; /* VT1720/4 - has SPDIF I/O */ unsigned int force_pdma4: 1; /* VT1720/4 - PDMA4 as non-spdif */ unsigned int force_rdma1: 1; /* VT1720/4 - RDMA1 as non-spdif */ + unsigned int midi_output: 1; /* VT1720/4: MIDI output triggered */ + unsigned int midi_input: 1; /* VT1720/4: MIDI input triggered */ unsigned int num_total_dacs; /* total DACs */ unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 67350901772c..e596d777d9dd 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -32,7 +32,7 @@ #include #include #include -#include +#include #include #include @@ -223,30 +223,153 @@ static unsigned int snd_vt1724_get_gpio_data(struct snd_ice1712 *ice) } /* - * MPU401 accessor + * MIDI */ -static unsigned char snd_vt1724_mpu401_read(struct snd_mpu401 *mpu, - unsigned long addr) + +static void vt1724_midi_clear_rx(struct snd_ice1712 *ice) +{ + unsigned int count; + + for (count = inb(ICEREG1724(ice, MPU_RXFIFO)); count > 0; --count) + inb(ICEREG1724(ice, MPU_DATA)); +} + +static inline struct snd_rawmidi_substream * +get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) { - /* fix status bits to the standard position */ - /* only RX_EMPTY and TX_FULL are checked */ - if (addr == MPU401C(mpu)) - return (inb(addr) & 0x0c) << 4; + return list_first_entry(&ice->rmidi[0]->streams[stream].substreams, + struct snd_rawmidi_substream, list); +} + +static void vt1724_midi_write(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_OUTPUT); + count = 31 - inb(ICEREG1724(ice, MPU_TXFIFO)); + if (count > 0) { + count = snd_rawmidi_transmit(s, buffer, count); + for (i = 0; i < count; ++i) + outb(buffer[i], ICEREG1724(ice, MPU_DATA)); + } +} + +static void vt1724_midi_read(struct snd_ice1712 *ice) +{ + struct snd_rawmidi_substream *s; + int count, i; + u8 buffer[32]; + + s = get_rawmidi_substream(ice, SNDRV_RAWMIDI_STREAM_INPUT); + count = inb(ICEREG1724(ice, MPU_RXFIFO)); + if (count > 0) { + count = min(count, 32); + for (i = 0; i < count; ++i) + buffer[i] = inb(ICEREG1724(ice, MPU_DATA)); + snd_rawmidi_receive(s, buffer, count); + } +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + u8 mask; + + spin_lock_irq(&ice->reg_lock); + mask = inb(ICEREG1724(ice, IRQMASK)); + if (enable) + mask &= ~flag; else - return inb(addr); + mask |= flag; + outb(mask, ICEREG1724(ice, IRQMASK)); + spin_unlock_irq(&ice->reg_lock); } -static void snd_vt1724_mpu401_write(struct snd_mpu401 *mpu, - unsigned char data, unsigned long addr) +static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) { - if (addr == MPU401C(mpu)) { - if (data == MPU401_ENTER_UART) - outb(0x01, addr); - /* what else? */ - } else - outb(data, addr); + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); + return 0; +} + +static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); + return 0; } +static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_output = 1; + vt1724_midi_write(ice); + } else { + ice->midi_output = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long timeout; + + /* 32 bytes should be transmitted in less than about 12 ms */ + timeout = jiffies + msecs_to_jiffies(15); + do { + if (inb(ICEREG1724(ice, MPU_CTRL)) & VT1724_MPU_TX_EMPTY) + break; + schedule_timeout_uninterruptible(1); + } while (time_after(timeout, jiffies)); +} + +static struct snd_rawmidi_ops vt1724_midi_output_ops = { + .open = vt1724_midi_output_open, + .close = vt1724_midi_output_close, + .trigger = vt1724_midi_output_trigger, + .drain = vt1724_midi_output_drain, +}; + +static int vt1724_midi_input_open(struct snd_rawmidi_substream *s) +{ + vt1724_midi_clear_rx(s->rmidi->private_data); + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 1); + return 0; +} + +static int vt1724_midi_input_close(struct snd_rawmidi_substream *s) +{ + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_RX, 0); + return 0; +} + +static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up) +{ + struct snd_ice1712 *ice = s->rmidi->private_data; + unsigned long flags; + + spin_lock_irqsave(&ice->reg_lock, flags); + if (up) { + ice->midi_input = 1; + vt1724_midi_read(ice); + } else { + ice->midi_input = 0; + } + spin_unlock_irqrestore(&ice->reg_lock, flags); +} + +static struct snd_rawmidi_ops vt1724_midi_input_ops = { + .open = vt1724_midi_input_open, + .close = vt1724_midi_input_close, + .trigger = vt1724_midi_input_trigger, +}; + /* * Interrupt handler @@ -278,13 +401,10 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) #endif handled = 1; if (status & VT1724_IRQ_MPU_TX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt_tx(irq, - ice->rmidi[0]->private_data); - else /* disable TX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_output) + vt1724_midi_write(ice); + spin_unlock(&ice->reg_lock); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -293,13 +413,12 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - if (ice->rmidi[0]) - snd_mpu401_uart_interrupt(irq, - ice->rmidi[0]->private_data); - else /* disable RX to be sure */ - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_RX, - ICEREG1724(ice, IRQMASK)); + spin_lock(&ice->reg_lock); + if (ice->midi_input) + vt1724_midi_read(ice); + else + vt1724_midi_clear_rx(ice); + spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); @@ -2425,28 +2544,30 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, if (! c->no_mpu401) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { - struct snd_mpu401 *mpu; - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, - ICEREG1724(ice, MPU_CTRL), - (MPU401_INFO_INTEGRATED | - MPU401_INFO_NO_ACK | - MPU401_INFO_TX_IRQ), - ice->irq, 0, - &ice->rmidi[0])) < 0) { + struct snd_rawmidi *rmidi; + + err = snd_rawmidi_new(card, "MIDI", 0, 1, 1, &rmidi); + if (err < 0) { snd_card_free(card); return err; } - mpu = ice->rmidi[0]->private_data; - mpu->read = snd_vt1724_mpu401_read; - mpu->write = snd_vt1724_mpu401_write; - /* unmask MPU RX/TX irqs */ - outb(inb(ICEREG1724(ice, IRQMASK)) & - ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX), - ICEREG1724(ice, IRQMASK)); + ice->rmidi[0] = rmidi; + rmidi->private_data = ice; + strcpy(rmidi->name, "ICE1724 MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &vt1724_midi_output_ops); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &vt1724_midi_input_ops); + /* set watermarks */ outb(VT1724_MPU_RX_FIFO | 0x1, ICEREG1724(ice, MPU_FIFO_WM)); outb(0x1, ICEREG1724(ice, MPU_FIFO_WM)); + /* set UART mode */ + outb(VT1724_MPU_UART, ICEREG1724(ice, MPU_CTRL)); } } -- cgit v1.2.3 From a72e72469a166c825196c3f20dabd352877fec2b Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 20 May 2008 01:06:55 +0300 Subject: [ALSA] remove CVS keywords This patch removes CVS keywords that weren't updated for a long time from comments. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_game.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/au88x0/au88x0_game.c b/sound/pci/au88x0/au88x0_game.c index bc212f41a38a..e291aa59742e 100644 --- a/sound/pci/au88x0/au88x0_game.c +++ b/sound/pci/au88x0/au88x0_game.c @@ -1,6 +1,4 @@ /* - * $Id: au88x0_game.c,v 1.9 2003/09/22 03:51:28 mjander Exp $ - * * Manuel Jander. * * Based on the work of: -- cgit v1.2.3 From 89fe5117928b2c1272c9376362131ded561c91ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:10:37 +0200 Subject: sound: Convert to menuconfig Convert menu in sound Kconfig files to menuconfig and if. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 81 ++++++++++--------------------------------------------- 1 file changed, 14 insertions(+), 67 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index d95fbb2b5b9f..b148c0b3ef3f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -1,11 +1,16 @@ # ALSA PCI drivers -menu "PCI devices" - depends on SND!=n && PCI +menuconfig SND_PCI + bool "PCI sound devices" + depends on PCI + default y + help + Support for sound devices connected via the PCI bus. + +if SND_PCI config SND_AD1889 tristate "Analog Devices AD1889" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -17,7 +22,6 @@ config SND_AD1889 config SND_ALS300 tristate "Avance Logic ALS300/ALS300+" - depends on SND select SND_PCM select SND_AC97_CODEC select SND_OPL3_LIB @@ -29,7 +33,7 @@ config SND_ALS300 config SND_ALS4000 tristate "Avance Logic ALS4000" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -43,7 +47,6 @@ config SND_ALS4000 config SND_ALI5451 tristate "ALi M5451 PCI Audio Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -57,7 +60,6 @@ config SND_ALI5451 config SND_ATIIXP tristate "ATI IXP AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -69,7 +71,6 @@ config SND_ATIIXP config SND_ATIIXP_MODEM tristate "ATI IXP Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -80,7 +81,6 @@ config SND_ATIIXP_MODEM config SND_AU8810 tristate "Aureal Advantage" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -95,7 +95,6 @@ config SND_AU8810 config SND_AU8820 tristate "Aureal Vortex" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -109,7 +108,6 @@ config SND_AU8820 config SND_AU8830 tristate "Aureal Vortex 2" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -124,7 +122,6 @@ config SND_AU8830 config SND_AW2 tristate "Emagic Audiowerk 2" - depends on SND help Say Y here to include support for Emagic Audiowerk 2 soundcards. @@ -139,7 +136,7 @@ config SND_AW2 config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on SND && EXPERIMENTAL + depends on EXPERIMENTAL select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -152,7 +149,6 @@ config SND_AZT3328 config SND_BT87X tristate "Bt87x Audio Capture" - depends on SND select SND_PCM help If you want to record audio from TV cards based on @@ -174,7 +170,6 @@ config SND_BT87X_OVERCLOCK config SND_CA0106 tristate "SB Audigy LS / Live 24bit" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI select SND_VMASTER @@ -187,7 +182,6 @@ config SND_CA0106 config SND_CMIPCI tristate "C-Media 8338, 8738, 8768, 8770" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -201,13 +195,11 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - depends on SND select SND_PCM select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -225,7 +217,6 @@ config SND_OXYGEN config SND_CS4281 tristate "Cirrus Logic (Sound Fusion) CS4281" - depends on SND select SND_OPL3_LIB select SND_RAWMIDI select SND_AC97_CODEC @@ -237,7 +228,6 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -258,7 +248,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" - depends on SND && ISA_DMA_API + depends on ISA_DMA_API select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. @@ -268,7 +258,7 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help @@ -286,7 +276,6 @@ config SND_CS5535AUDIO config SND_DARLA20 tristate "(Echoaudio) Darla20" - depends on SND select FW_LOADER select SND_PCM help @@ -297,7 +286,6 @@ config SND_DARLA20 config SND_GINA20 tristate "(Echoaudio) Gina20" - depends on SND select FW_LOADER select SND_PCM help @@ -308,7 +296,6 @@ config SND_GINA20 config SND_LAYLA20 tristate "(Echoaudio) Layla20" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -320,7 +307,6 @@ config SND_LAYLA20 config SND_DARLA24 tristate "(Echoaudio) Darla24" - depends on SND select FW_LOADER select SND_PCM help @@ -331,7 +317,6 @@ config SND_DARLA24 config SND_GINA24 tristate "(Echoaudio) Gina24" - depends on SND select FW_LOADER select SND_PCM help @@ -342,7 +327,6 @@ config SND_GINA24 config SND_LAYLA24 tristate "(Echoaudio) Layla24" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -354,7 +338,6 @@ config SND_LAYLA24 config SND_MONA tristate "(Echoaudio) Mona" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -366,7 +349,6 @@ config SND_MONA config SND_MIA tristate "(Echoaudio) Mia" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -378,7 +360,6 @@ config SND_MIA config SND_ECHO3G tristate "(Echoaudio) 3G cards" - depends on SND select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -390,7 +371,6 @@ config SND_ECHO3G config SND_INDIGO tristate "(Echoaudio) Indigo" - depends on SND select FW_LOADER select SND_PCM help @@ -401,7 +381,6 @@ config SND_INDIGO config SND_INDIGOIO tristate "(Echoaudio) Indigo IO" - depends on SND select FW_LOADER select SND_PCM help @@ -412,7 +391,6 @@ config SND_INDIGOIO config SND_INDIGODJ tristate "(Echoaudio) Indigo DJ" - depends on SND select FW_LOADER select SND_PCM help @@ -423,7 +401,6 @@ config SND_INDIGODJ config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" - depends on SND select FW_LOADER select SND_HWDEP select SND_RAWMIDI @@ -441,7 +418,6 @@ config SND_EMU10K1 config SND_EMU10K1X tristate "Emu10k1X (Dell OEM Version)" - depends on SND select SND_AC97_CODEC select SND_RAWMIDI help @@ -453,7 +429,6 @@ config SND_EMU10K1X config SND_ENS1370 tristate "(Creative) Ensoniq AudioPCI 1370" - depends on SND select SND_RAWMIDI select SND_PCM help @@ -464,7 +439,6 @@ config SND_ENS1370 config SND_ENS1371 tristate "(Creative) Ensoniq AudioPCI 1371/1373" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC help @@ -476,7 +450,6 @@ config SND_ENS1371 config SND_ES1938 tristate "ESS ES1938/1946/1969 (Solo-1)" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -489,7 +462,6 @@ config SND_ES1938 config SND_ES1968 tristate "ESS ES1968/1978 (Maestro-1/2/2E)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -501,7 +473,6 @@ config SND_ES1968 config SND_FM801 tristate "ForteMedia FM801" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -528,7 +499,6 @@ config SND_FM801_TEA575X config SND_HDA_INTEL tristate "Intel HD Audio" - depends on SND select SND_PCM select SND_VMASTER help @@ -637,7 +607,6 @@ config SND_HDA_POWER_SAVE_DEFAULT config SND_HDSP tristate "RME Hammerfall DSP Audio" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -650,7 +619,6 @@ config SND_HDSP config SND_HDSPM tristate "RME Hammerfall DSP MADI" - depends on SND select SND_HWDEP select SND_RAWMIDI select SND_PCM @@ -663,7 +631,6 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for the MediaTek/TempoTec HiFier @@ -674,7 +641,6 @@ config SND_HIFIER config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -691,7 +657,6 @@ config SND_ICE1712 config SND_ICE1724 tristate "ICE/VT1724/1720 (Envy24HT/PT)" - depends on SND select SND_RAWMIDI select SND_AC97_CODEC select SND_VMASTER @@ -709,7 +674,6 @@ config SND_ICE1724 config SND_INTEL8X0 tristate "Intel/SiS/nVidia/AMD/ALi AC97 Controller" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated AC97 sound @@ -722,7 +686,6 @@ config SND_INTEL8X0 config SND_INTEL8X0M tristate "Intel/SiS/nVidia/AMD MC97 Modem" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -733,7 +696,6 @@ config SND_INTEL8X0M config SND_KORG1212 tristate "Korg 1212 IO" - depends on SND select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL select SND_PCM help @@ -753,7 +715,6 @@ config SND_KORG1212_FIRMWARE_IN_KERNEL config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" - depends on SND select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL select SND_AC97_CODEC help @@ -774,7 +735,6 @@ config SND_MAESTRO3_FIRMWARE_IN_KERNEL config SND_MIXART tristate "Digigram miXart" - depends on SND select SND_HWDEP select SND_PCM help @@ -786,7 +746,6 @@ config SND_MIXART config SND_NM256 tristate "NeoMagic NM256AV/ZX" - depends on SND select SND_AC97_CODEC help Say Y here to include support for NeoMagic NM256AV/ZX chips. @@ -796,7 +755,6 @@ config SND_NM256 config SND_PCXHR tristate "Digigram PCXHR" - depends on SND select SND_PCM select SND_HWDEP help @@ -807,7 +765,6 @@ config SND_PCXHR config SND_RIPTIDE tristate "Conexant Riptide" - depends on SND select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART @@ -820,7 +777,6 @@ config SND_RIPTIDE config SND_RME32 tristate "RME Digi32, 32/8, 32 PRO" - depends on SND select SND_PCM help Say Y to include support for RME Digi32, Digi32 PRO and @@ -832,7 +788,6 @@ config SND_RME32 config SND_RME96 tristate "RME Digi96, 96/8, 96/8 PRO" - depends on SND select SND_PCM help Say Y here to include support for RME Digi96, Digi96/8 and @@ -843,7 +798,6 @@ config SND_RME96 config SND_RME9652 tristate "RME Digi9652 (Hammerfall)" - depends on SND select SND_PCM help Say Y here to include support for RME Hammerfall (RME @@ -854,7 +808,7 @@ config SND_RME9652 config SND_SIS7019 tristate "SiS 7019 Audio Accelerator" - depends on SND && X86 && !X86_64 + depends on X86 && !X86_64 select SND_AC97_CODEC help Say Y here to include support for the SiS 7019 Audio Accelerator. @@ -864,7 +818,6 @@ config SND_SIS7019 config SND_SONICVIBES tristate "S3 SonicVibes" - depends on SND select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -877,7 +830,6 @@ config SND_SONICVIBES config SND_TRIDENT tristate "Trident 4D-Wave DX/NX; SiS 7018" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -889,7 +841,6 @@ config SND_TRIDENT config SND_VIA82XX tristate "VIA 82C686A/B, 8233/8235 AC97 Controller" - depends on SND select SND_MPU401_UART select SND_AC97_CODEC help @@ -901,7 +852,6 @@ config SND_VIA82XX config SND_VIA82XX_MODEM tristate "VIA 82C686A/B, 8233 based Modems" - depends on SND select SND_AC97_CODEC help Say Y here to include support for the integrated MC97 modem on @@ -912,7 +862,6 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 100/200 (Xonar)" - depends on SND select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the @@ -923,7 +872,6 @@ config SND_VIRTUOSO config SND_VX222 tristate "Digigram VX222" - depends on SND select SND_VX_LIB help Say Y here to include support for Digigram VX222 soundcards. @@ -933,7 +881,6 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" - depends on SND select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL select SND_OPL3_LIB select SND_MPU401_UART @@ -975,4 +922,4 @@ 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. -endmenu +endif # SND_PCI -- cgit v1.2.3 From 6938d6b2a90d5e2ffaef037852652a1333502519 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:11:26 +0200 Subject: [ALSA] Fix AC97 kconfig items The kconfig items related with AC97-powersave must be outside the CONFIG_SND_PCI range. And it'd be better together with CONFIG_SND_AC97. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 21 --------------------- 1 file changed, 21 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index b148c0b3ef3f..8fe5dac39428 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -901,25 +901,4 @@ config SND_YMFPCI_FIRMWARE_IN_KERNEL for the YMFPCI driver. If you choose N here, you need to install the firmware files from the alsa-firmware package. -config SND_AC97_POWER_SAVE - bool "AC97 Power-Saving Mode" - depends on SND_AC97_CODEC && EXPERIMENTAL - default n - help - Say Y here to enable the aggressive power-saving support of - AC97 codecs. In this mode, the power-mode is dynamically - controlled at each open/close. - - The mode is activated by passing power_save=1 option to - snd-ac97-codec driver. You can toggle it dynamically over - sysfs, too. - -config SND_AC97_POWER_SAVE_DEFAULT - int "Default time-out for AC97 power-save mode" - depends on SND_AC97_POWER_SAVE - default 0 - help - The default time-out value in seconds for AC97 automatic - power-save mode. 0 means to disable the power-save mode. - endif # SND_PCI -- cgit v1.2.3 From 62cf872a8eec1f11aacbec0ac3fe3698bfa9b403 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 May 2008 12:15:15 +0200 Subject: [ALSA] Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE to represent its meaning more better. This config isn't provided only for the detection but for more verbose debug prints in general. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/pcxhr/pcxhr.c | 4 ++-- sound/pci/pcxhr/pcxhr_core.c | 18 +++++++++--------- 4 files changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a6be6e3e8716..d2e1093f8e97 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2335,7 +2335,7 @@ int snd_hda_check_board_config(struct hda_codec *codec, if (!tbl) return -1; if (tbl->value >= 0 && tbl->value < num_configs) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE char tmp[10]; const char *model = NULL; if (models) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 2177d9af5334..6e18a422d993 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -88,7 +88,7 @@ static int hda_hwdep_ioctl_compat(struct snd_hwdep *hw, struct file *file, static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) { -#ifndef CONFIG_SND_DEBUG_DETECT +#ifndef CONFIG_SND_DEBUG_VERBOSE if (!capable(CAP_SYS_RAWIO)) return -EACCES; #endif diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 7fdcdc8c6b64..2c7e25336795 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -516,7 +516,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) int capture_mask = 0; int playback_mask = 0; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -623,7 +623,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) mutex_unlock(&mgr->setup_mutex); -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***TRIGGER TASKLET*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 78aa81feaa4a..abe5c59b72df 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -473,7 +473,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, }; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* cmd_names[] = { [CMD_VERSION] = "CMD_VERSION", [CMD_SUPPORTED] = "CMD_SUPPORTED", @@ -549,7 +549,7 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) } } } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" stat[%d]=%x\n", i, data); #endif @@ -597,7 +597,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data |= 0x008000; /* MASK_MORE_THAN_1_WORD_COMMAND */ else data &= 0xff7fff; /* MASK_1_WORD_COMMAND */ -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]); #endif @@ -624,7 +624,7 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) for (i=1; i < rmh->cmd_len; i++) { /* send other words */ data = rmh->cmd[i]; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) snd_printdd(" cmd[%d]=%x\n", i, data); #endif @@ -847,7 +847,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m int state, i, err; int audio_mask; -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif @@ -894,7 +894,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m if (err) return err; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE do_gettimeofday(&my_tv2); snd_printdd("***SET PIPE STATE*** TIME = %ld (err = %x)\n", (long)(my_tv2.tv_usec - my_tv1.tv_usec), err); @@ -951,7 +951,7 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, enum pcxhr_async_err_src err_src, int pipe, int is_capture) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE static char* err_src_name[] = { [PCXHR_ERR_PIPE] = "Pipe", [PCXHR_ERR_STREAM] = "Stream", @@ -1169,7 +1169,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->dsp_time_last, dsp_time_new); mgr->dsp_time_err++; } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new); else if (dsp_time_diff >= (2*PCXHR_GRANULARITY)) @@ -1208,7 +1208,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) mgr->src_it_dsp = reg; tasklet_hi_schedule(&mgr->msg_taskq); } -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE if (reg & PCXHR_FATAL_DSP_ERR) snd_printdd("FATAL DSP ERROR : %x\n", reg); #endif -- cgit v1.2.3 From f269002e61446ed3410d8ca5f06ebca1e2760cb5 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 11:44:55 +0200 Subject: [ALSA] hda - Add support of Teradici controller Add the new PCI ID 0x6549 0x1200 Teradici controller. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6ba7ac01d9f6..7f62196989a1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -197,6 +197,10 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define ATIHDMI_NUM_CAPTURE 0 #define ATIHDMI_NUM_PLAYBACK 1 +/* TERA has 4 playback and 3 capture */ +#define TERA_NUM_CAPTURE 3 +#define TERA_NUM_PLAYBACK 4 + /* this number is statically defined for simplicity */ #define MAX_AZX_DEV 16 @@ -384,6 +388,7 @@ enum { AZX_DRIVER_SIS, AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, + AZX_DRIVER_TERA, }; static char *driver_short_names[] __devinitdata = { @@ -395,6 +400,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", + [AZX_DRIVER_TERA] = "HDA Teradici", }; /* @@ -1106,6 +1112,7 @@ static unsigned int azx_max_codecs[] __devinitdata = { [AZX_DRIVER_SIS] = 3, /* FIXME: correct? */ [AZX_DRIVER_ULI] = 3, /* FIXME: correct? */ [AZX_DRIVER_NVIDIA] = 3, /* FIXME: correct? */ + [AZX_DRIVER_TERA] = 1, }; static int __devinit azx_codec_create(struct azx *chip, const char *model, @@ -2229,6 +2236,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + /* Teradici */ + { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); -- cgit v1.2.3 From abbc9d1b25637b1948a4718fa8f7b257233136bc Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 11:48:01 +0200 Subject: [ALSA] hda - Add ICH9 controller support (8086:2911) Added the missing PCI ID for ICH9 controller (8086:2911) Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f62196989a1..6e2dc4f5ca7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2184,6 +2184,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, + { PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, -- cgit v1.2.3 From 6dda9f4a95905f2b38e79e3737da5e25397e6acb Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:05:31 +0200 Subject: [ALSA] hda - Add ALC663 support Added the support of ALC663 codec, including specific models for ASUS M51VA, ASUS G71V, ASUS H13 and ASUS G50V. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 289 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 286 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b0a2a262ece2..c659588e26d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -163,6 +163,10 @@ enum { ALC662_LENOVO_101E, ALC662_ASUS_EEEPC_P701, ALC662_ASUS_EEEPC_EP20, + ALC663_ASUS_M51VA, + ALC663_ASUS_G71V, + ALC663_ASUS_H13, + ALC663_ASUS_G50V, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -13251,6 +13255,23 @@ static struct hda_input_mux alc662_eeepc_capture_source = { }, }; +static struct hda_input_mux alc663_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc663_m51va_capture_source = { + .num_items = 2, + .items = { + { "Ext-Mic", 0x0 }, + { "D-Mic", 0x9 }, + }, +}; + #define alc662_mux_enum_info alc_mux_enum_info #define alc662_mux_enum_get alc_mux_enum_get #define alc662_mux_enum_put alc882_mux_enum_put @@ -13431,6 +13452,44 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc663_m51va_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g71v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_g50v_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -13501,6 +13560,11 @@ static struct hda_verb alc662_init_verbs[] = { {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* always trun on EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } }; @@ -13571,6 +13635,43 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; +static struct hda_verb alc663_m51va_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g71v_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ + /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */ + + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_g50v_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc662_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), @@ -13692,6 +13793,125 @@ static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) alc662_eeepc_ep20_automute(codec); } +static void alc663_m51va_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_m51va_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); +} + +static void alc663_m51va_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc663_m51va_mic_automute(codec); + break; + } +} + +static void alc663_m51va_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc663_m51va_mic_automute(codec); +} + +static void alc663_g71v_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_front_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc663_g71v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_g71v_hp_automute(codec); + break; + case ALC880_FRONT_EVENT: + alc663_g71v_front_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g71v_inithook(struct hda_codec *codec) +{ + alc663_g71v_front_automute(codec); + alc663_g71v_hp_automute(codec); + alc662_eeepc_mic_automute(codec); +} + +static void alc663_g50v_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_g50v_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -13714,14 +13934,24 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", + [ALC663_ASUS_M51VA] = "m51va", + [ALC663_ASUS_G71V] = "g71v", + [ALC663_ASUS_H13] = "h13", + [ALC663_ASUS_G50V] = "g50v", [ALC662_AUTO] = "auto", }; static struct snd_pci_quirk alc662_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V), SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), {} }; @@ -13809,7 +14039,53 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_eeepc_ep20_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, - + [ALC663_ASUS_M51VA] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G71V] = { + .mixers = { alc663_g71v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_g71v_unsol_event, + .init_hook = alc663_g71v_inithook, + }, + [ALC663_ASUS_H13] = { + .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc663_m51va_capture_source, + .unsol_event = alc663_m51va_unsol_event, + .init_hook = alc663_m51va_inithook, + }, + [ALC663_ASUS_G50V] = { + .mixers = { alc663_g50v_mixer, alc662_capture_mixer}, + .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), + .channel_mode = alc662_3ST_6ch_modes, + .input_mux = &alc663_capture_source, + .unsol_event = alc663_g50v_unsol_event, + .init_hook = alc663_g50v_inithook, + }, }; @@ -14108,11 +14384,17 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); - spec->stream_name_analog = "ALC662 Analog"; + if (codec->vendor_id == 0x10ec0663) { + spec->stream_name_analog = "ALC663 Analog"; + spec->stream_name_digital = "ALC663 Digital"; + } else { + spec->stream_name_analog = "ALC662 Analog"; + spec->stream_name_digital = "ALC662 Digital"; + } + spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; - spec->stream_name_digital = "ALC662 Digital"; spec->stream_digital_playback = &alc662_pcm_digital_playback; spec->stream_digital_capture = &alc662_pcm_digital_capture; @@ -14151,6 +14433,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc883 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, -- cgit v1.2.3 From 531240ff520406c793a110e1c0f187d931f47d66 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:10:25 +0200 Subject: [ALSA] hda - Fix vref pincap check in alc882 auto-detection Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c659588e26d3..8174050da15f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6357,7 +6357,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) continue; vref = PIN_IN; if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - if (snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) & + unsigned int pincap; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if ((pincap >> AC_PINCAP_VREF_SHIFT) & AC_PINCAP_VREF_80) vref = PIN_VREF80; } -- cgit v1.2.3 From 2f8932863d243a744ccd3dc005490ad9d2eae478 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:14:47 +0200 Subject: [ALSA] hda - show correct codec chip in PCM stream names Show more exact codec chip name in the PCM stream name strings. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8174050da15f..5770b9c3efa2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6491,14 +6491,20 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - spec->stream_name_analog = "ALC882 Analog"; + if (codec->vendor_id == 0x10ec0885) { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } else { + spec->stream_name_analog = "ALC882 Analog"; + spec->stream_name_digital = "ALC882 Digital"; + } + spec->stream_analog_playback = &alc882_pcm_analog_playback; spec->stream_analog_capture = &alc882_pcm_analog_capture; /* FIXME: setup DAC5 */ /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC882 Digital"; spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; @@ -8177,12 +8183,25 @@ static int patch_alc883(struct hda_codec *codec) if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); - spec->stream_name_analog = "ALC883 Analog"; + switch (codec->vendor_id) { + case 0x10ec0888: + spec->stream_name_analog = "ALC888 Analog"; + spec->stream_name_digital = "ALC888 Digital"; + break; + case 0x10ec0889: + spec->stream_name_analog = "ALC889 Analog"; + spec->stream_name_digital = "ALC889 Digital"; + break; + default: + spec->stream_name_analog = "ALC883 Analog"; + spec->stream_name_digital = "ALC883 Digital"; + break; + } + spec->stream_analog_playback = &alc883_pcm_analog_playback; spec->stream_analog_capture = &alc883_pcm_analog_capture; spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC883 Digital"; spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; @@ -10680,12 +10699,18 @@ static int patch_alc268(struct hda_codec *codec) if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); - spec->stream_name_analog = "ALC268 Analog"; + if (codec->vendor_id == 0x10ec0267) { + spec->stream_name_analog = "ALC267 Analog"; + spec->stream_name_digital = "ALC267 Digital"; + } else { + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_name_digital = "ALC268 Digital"; + } + spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture; - spec->stream_name_digital = "ALC268 Digital"; spec->stream_digital_playback = &alc268_pcm_digital_playback; if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) @@ -13174,11 +13199,17 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); - spec->stream_name_analog = "ALC861VD Analog"; + if (codec->vendor_id == 0x10ec0660) { + spec->stream_name_analog = "ALC660-VD Analog"; + spec->stream_name_digital = "ALC660-VD Digital"; + } else { + spec->stream_name_analog = "ALC861VD Analog"; + spec->stream_name_digital = "ALC861VD Digital"; + } + spec->stream_analog_playback = &alc861vd_pcm_analog_playback; spec->stream_analog_capture = &alc861vd_pcm_analog_capture; - spec->stream_name_digital = "ALC861VD Digital"; spec->stream_digital_playback = &alc861vd_pcm_digital_playback; spec->stream_digital_capture = &alc861vd_pcm_digital_capture; -- cgit v1.2.3 From f9423e7a94eb2dfef3503dde76d17eaf342ab962 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 27 May 2008 12:32:25 +0200 Subject: [ALSA] hda - Fix EAPD and COEF setups for realtek codecs Fixed EAPD and COEF setups for Realtek ALC662/663, 660-VD and 888 codecs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++++++++++++++++-- 1 file changed, 35 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5770b9c3efa2..cb3e0283f0ce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -780,6 +780,24 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) alc_sku_automute(codec); } +/* additional initialization for ALC888 variants */ +static void alc888_coef_init(struct hda_codec *codec) +{ + unsigned int tmp; + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); + tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); + if ((tmp & 0xf0) == 2) + /* alc888S-VC */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x830); + else + /* alc888-VB */ + snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x3030); +} + /* 32-bit subsystem ID for BIOS loading in HD Audio codec. * 31 ~ 16 : Manufacture ID * 15 ~ 8 : SKU ID @@ -855,8 +873,10 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: case 0x10ec0862: - case 0x10ec0662: case 0x10ec0889: snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_EAPD_BTLENABLE, 2); @@ -881,7 +901,6 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: - case 0x10ec0888: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -893,6 +912,9 @@ do_sku: AC_VERB_SET_PROC_COEF, tmp | 0x2010); break; + case 0x10ec0888: + alc888_coef_init(codec); + break; case 0x10ec0267: case 0x10ec0268: snd_hda_codec_write(codec, 0x20, 0, @@ -8214,6 +8236,9 @@ static int patch_alc883(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; + else if (codec->vendor_id == 0x10ec0888) + spec->init_hook = alc888_coef_init; + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc883_loopbacks; @@ -12662,6 +12687,12 @@ static struct hda_verb alc861vd_eapd_verbs[] = { { } }; +static struct hda_verb alc660vd_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -13202,6 +13233,8 @@ static int patch_alc861vd(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0660) { spec->stream_name_analog = "ALC660-VD Analog"; spec->stream_name_digital = "ALC660-VD Digital"; + /* always turn on EAPD */ + spec->init_verbs[spec->num_init_verbs++] = alc660vd_eapd_verbs; } else { spec->stream_name_analog = "ALC861VD Analog"; spec->stream_name_digital = "ALC861VD Digital"; -- cgit v1.2.3 From c18f68e4d809a517ed8df540bac2993a4f14d9a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 May 2008 18:53:05 +0200 Subject: [ALSA] Clean up sound/pci/ac97/Makefile Signed-off-by: Takashi Iwai --- sound/pci/ac97/Makefile | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index 0be48b1a22d0..ae36950f2568 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -3,16 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ac97-codec-objs := ac97_codec.o ac97_pcm.o - -ifneq ($(CONFIG_PROC_FS),) -snd-ac97-codec-objs += ac97_proc.o -endif +snd-ac97-codec-y := ac97_codec.o ac97_pcm.o +snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o snd-ak4531-codec-objs := ak4531_codec.o # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o - -obj-m := $(sort $(obj-m)) -- cgit v1.2.3 From 23ce1547638443f0053dd674e728062c48ff0e39 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:22:22 +0200 Subject: [ALSA] Make ak4531 local to ens1370 driver The ak4531 module is used only by ens1370 driver (and unlikely that any other will use it ever). Let's make it local to ens1370. Signed-off-by: Takashi Iwai --- sound/pci/Makefile | 2 +- sound/pci/ac97/Makefile | 3 - sound/pci/ac97/ak4531_codec.c | 508 ------------------------------------------ sound/pci/ak4531_codec.c | 488 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 489 insertions(+), 512 deletions(-) delete mode 100644 sound/pci/ac97/ak4531_codec.c create mode 100644 sound/pci/ak4531_codec.c (limited to 'sound/pci') diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 85ef14bc8056..65b25d221cd2 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -13,7 +13,7 @@ snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o snd-cs5530-objs := cs5530.o -snd-ens1370-objs := ens1370.o +snd-ens1370-objs := ens1370.o ak4531_codec.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o snd-es1968-objs := es1968.o diff --git a/sound/pci/ac97/Makefile b/sound/pci/ac97/Makefile index ae36950f2568..41fa322f0971 100644 --- a/sound/pci/ac97/Makefile +++ b/sound/pci/ac97/Makefile @@ -6,8 +6,5 @@ snd-ac97-codec-y := ac97_codec.o ac97_pcm.o snd-ac97-codec-$(CONFIG_PROC_FS) += ac97_proc.o -snd-ak4531-codec-objs := ak4531_codec.o - # Toplevel Module Dependency obj-$(CONFIG_SND_AC97_CODEC) += snd-ac97-codec.o -obj-$(CONFIG_SND_ENS1370) += snd-ak4531-codec.o diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c deleted file mode 100644 index c0c1633999ea..000000000000 --- a/sound/pci/ac97/ak4531_codec.c +++ /dev/null @@ -1,508 +0,0 @@ -/* - * Copyright (c) by Jaroslav Kysela - * Universal routines for AK4531 codec - * - * - * 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 -#include -#include -#include - -#include -#include -#include - -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Universal routines for AK4531 codec"); -MODULE_LICENSE("GPL"); - -#ifdef CONFIG_PROC_FS -static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531); -#else -#define snd_ak4531_proc_init(card,ak) -#endif - -/* - * - */ - -#if 0 - -static void snd_ak4531_dump(struct snd_ak4531 *ak4531) -{ - int idx; - - for (idx = 0; idx < 0x19; idx++) - printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); -} - -#endif - -/* - * - */ - -#define AK4531_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_ak4531_info_single, \ - .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ - .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22) } -#define AK4531_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, .index = xindex, \ - .info = snd_ak4531_info_single, \ - .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ - .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22), \ - .tlv = { .p = (xtlv) } } - -static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ak4531_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 16) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int val; - - mutex_lock(&ak4531->reg_mutex); - val = (ak4531->regs[reg] >> shift) & mask; - mutex_unlock(&ak4531->reg_mutex); - if (invert) { - val = mask - val; - } - ucontrol->value.integer.value[0] = val; - return 0; -} - -static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 16) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - int val; - - val = ucontrol->value.integer.value[0] & mask; - if (invert) { - val = mask - val; - } - val <<= shift; - mutex_lock(&ak4531->reg_mutex); - val = (ak4531->regs[reg] & ~(mask << shift)) | val; - change = val != ak4531->regs[reg]; - ak4531->write(ak4531, reg, ak4531->regs[reg] = val); - mutex_unlock(&ak4531->reg_mutex); - return change; -} - -#define AK4531_DOUBLE(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_ak4531_info_double, \ - .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ - .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22) } -#define AK4531_DOUBLE_TLV(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, .index = xindex, \ - .info = snd_ak4531_info_double, \ - .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ - .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22), \ - .tlv = { .p = (xtlv) } } - -static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ak4531_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int left_shift = (kcontrol->private_value >> 16) & 0x07; - int right_shift = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int left, right; - - mutex_lock(&ak4531->reg_mutex); - left = (ak4531->regs[left_reg] >> left_shift) & mask; - right = (ak4531->regs[right_reg] >> right_shift) & mask; - mutex_unlock(&ak4531->reg_mutex); - if (invert) { - left = mask - left; - right = mask - right; - } - ucontrol->value.integer.value[0] = left; - ucontrol->value.integer.value[1] = right; - return 0; -} - -static int snd_ak4531_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int left_shift = (kcontrol->private_value >> 16) & 0x07; - int right_shift = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - int left, right; - - left = ucontrol->value.integer.value[0] & mask; - right = ucontrol->value.integer.value[1] & mask; - if (invert) { - left = mask - left; - right = mask - right; - } - left <<= left_shift; - right <<= right_shift; - mutex_lock(&ak4531->reg_mutex); - if (left_reg == right_reg) { - left = (ak4531->regs[left_reg] & ~((mask << left_shift) | (mask << right_shift))) | left | right; - change = left != ak4531->regs[left_reg]; - ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left); - } else { - left = (ak4531->regs[left_reg] & ~(mask << left_shift)) | left; - right = (ak4531->regs[right_reg] & ~(mask << right_shift)) | right; - change = left != ak4531->regs[left_reg] || right != ak4531->regs[right_reg]; - ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left); - ak4531->write(ak4531, right_reg, ak4531->regs[right_reg] = right); - } - mutex_unlock(&ak4531->reg_mutex); - return change; -} - -#define AK4531_INPUT_SW(xname, xindex, reg1, reg2, left_shift, right_shift) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_ak4531_info_input_sw, \ - .get = snd_ak4531_get_input_sw, .put = snd_ak4531_put_input_sw, \ - .private_value = reg1 | (reg2 << 8) | (left_shift << 16) | (right_shift << 24) } - -static int snd_ak4531_info_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 4; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int snd_ak4531_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int reg1 = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 8) & 0xff; - int left_shift = (kcontrol->private_value >> 16) & 0x0f; - int right_shift = (kcontrol->private_value >> 24) & 0x0f; - - mutex_lock(&ak4531->reg_mutex); - ucontrol->value.integer.value[0] = (ak4531->regs[reg1] >> left_shift) & 1; - ucontrol->value.integer.value[1] = (ak4531->regs[reg2] >> left_shift) & 1; - ucontrol->value.integer.value[2] = (ak4531->regs[reg1] >> right_shift) & 1; - ucontrol->value.integer.value[3] = (ak4531->regs[reg2] >> right_shift) & 1; - mutex_unlock(&ak4531->reg_mutex); - return 0; -} - -static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); - int reg1 = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 8) & 0xff; - int left_shift = (kcontrol->private_value >> 16) & 0x0f; - int right_shift = (kcontrol->private_value >> 24) & 0x0f; - int change; - int val1, val2; - - mutex_lock(&ak4531->reg_mutex); - val1 = ak4531->regs[reg1] & ~((1 << left_shift) | (1 << right_shift)); - val2 = ak4531->regs[reg2] & ~((1 << left_shift) | (1 << right_shift)); - val1 |= (ucontrol->value.integer.value[0] & 1) << left_shift; - val2 |= (ucontrol->value.integer.value[1] & 1) << left_shift; - val1 |= (ucontrol->value.integer.value[2] & 1) << right_shift; - val2 |= (ucontrol->value.integer.value[3] & 1) << right_shift; - change = val1 != ak4531->regs[reg1] || val2 != ak4531->regs[reg2]; - ak4531->write(ak4531, reg1, ak4531->regs[reg1] = val1); - ak4531->write(ak4531, reg2, ak4531->regs[reg2] = val2); - mutex_unlock(&ak4531->reg_mutex); - return change; -} - -static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0); -static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0); -static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0); - -static struct snd_kcontrol_new snd_ak4531_controls[] = { - -AK4531_DOUBLE_TLV("Master Playback Switch", 0, - AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1, - db_scale_master), -AK4531_DOUBLE("Master Playback Volume", 0, AK4531_LMASTER, AK4531_RMASTER, 0, 0, 0x1f, 1), - -AK4531_SINGLE_TLV("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1, - db_scale_mono), -AK4531_SINGLE("Master Mono Playback Volume", 0, AK4531_MONO_OUT, 0, 0x07, 1), - -AK4531_DOUBLE("PCM Switch", 0, AK4531_LVOICE, AK4531_RVOICE, 7, 7, 1, 1), -AK4531_DOUBLE_TLV("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1, - db_scale_input), -AK4531_DOUBLE("PCM Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 3, 2, 1, 0), -AK4531_DOUBLE("PCM Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 2, 2, 1, 0), - -AK4531_DOUBLE("PCM Switch", 1, AK4531_LFM, AK4531_RFM, 7, 7, 1, 1), -AK4531_DOUBLE_TLV("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1, - db_scale_input), -AK4531_DOUBLE("PCM Playback Switch", 1, AK4531_OUT_SW1, AK4531_OUT_SW1, 6, 5, 1, 0), -AK4531_INPUT_SW("PCM Capture Route", 1, AK4531_LIN_SW1, AK4531_RIN_SW1, 6, 5), - -AK4531_DOUBLE("CD Switch", 0, AK4531_LCD, AK4531_RCD, 7, 7, 1, 1), -AK4531_DOUBLE_TLV("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1, - db_scale_input), -AK4531_DOUBLE("CD Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 2, 1, 1, 0), -AK4531_INPUT_SW("CD Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 2, 1), - -AK4531_DOUBLE("Line Switch", 0, AK4531_LLINE, AK4531_RLINE, 7, 7, 1, 1), -AK4531_DOUBLE_TLV("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1, - db_scale_input), -AK4531_DOUBLE("Line Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 4, 3, 1, 0), -AK4531_INPUT_SW("Line Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 4, 3), - -AK4531_DOUBLE("Aux Switch", 0, AK4531_LAUXA, AK4531_RAUXA, 7, 7, 1, 1), -AK4531_DOUBLE_TLV("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1, - db_scale_input), -AK4531_DOUBLE("Aux Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 5, 4, 1, 0), -AK4531_INPUT_SW("Aux Capture Route", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 4, 3), - -AK4531_SINGLE("Mono Switch", 0, AK4531_MONO1, 7, 1, 1), -AK4531_SINGLE_TLV("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1, db_scale_input), -AK4531_SINGLE("Mono Playback Switch", 0, AK4531_OUT_SW2, 0, 1, 0), -AK4531_DOUBLE("Mono Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 0, 0, 1, 0), - -AK4531_SINGLE("Mono Switch", 1, AK4531_MONO2, 7, 1, 1), -AK4531_SINGLE_TLV("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1, db_scale_input), -AK4531_SINGLE("Mono Playback Switch", 1, AK4531_OUT_SW2, 1, 1, 0), -AK4531_DOUBLE("Mono Capture Switch", 1, AK4531_LIN_SW2, AK4531_RIN_SW2, 1, 1, 1, 0), - -AK4531_SINGLE_TLV("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1, db_scale_input), -AK4531_SINGLE("Mic Switch", 0, AK4531_MIC, 7, 1, 1), -AK4531_SINGLE("Mic Playback Switch", 0, AK4531_OUT_SW1, 0, 1, 0), -AK4531_DOUBLE("Mic Capture Switch", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 0, 0, 1, 0), - -AK4531_DOUBLE("Mic Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 7, 7, 1, 0), -AK4531_DOUBLE("Mono1 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 6, 6, 1, 0), -AK4531_DOUBLE("Mono2 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 5, 5, 1, 0), - -AK4531_SINGLE("AD Input Select", 0, AK4531_AD_IN, 0, 1, 0), -AK4531_SINGLE("Mic Boost (+30dB)", 0, AK4531_MIC_GAIN, 0, 1, 0) -}; - -static int snd_ak4531_free(struct snd_ak4531 *ak4531) -{ - if (ak4531) { - if (ak4531->private_free) - ak4531->private_free(ak4531); - kfree(ak4531); - } - return 0; -} - -static int snd_ak4531_dev_free(struct snd_device *device) -{ - struct snd_ak4531 *ak4531 = device->device_data; - return snd_ak4531_free(ak4531); -} - -static u8 snd_ak4531_initial_map[0x19 + 1] = { - 0x9f, /* 00: Master Volume Lch */ - 0x9f, /* 01: Master Volume Rch */ - 0x9f, /* 02: Voice Volume Lch */ - 0x9f, /* 03: Voice Volume Rch */ - 0x9f, /* 04: FM Volume Lch */ - 0x9f, /* 05: FM Volume Rch */ - 0x9f, /* 06: CD Audio Volume Lch */ - 0x9f, /* 07: CD Audio Volume Rch */ - 0x9f, /* 08: Line Volume Lch */ - 0x9f, /* 09: Line Volume Rch */ - 0x9f, /* 0a: Aux Volume Lch */ - 0x9f, /* 0b: Aux Volume Rch */ - 0x9f, /* 0c: Mono1 Volume */ - 0x9f, /* 0d: Mono2 Volume */ - 0x9f, /* 0e: Mic Volume */ - 0x87, /* 0f: Mono-out Volume */ - 0x00, /* 10: Output Mixer SW1 */ - 0x00, /* 11: Output Mixer SW2 */ - 0x00, /* 12: Lch Input Mixer SW1 */ - 0x00, /* 13: Rch Input Mixer SW1 */ - 0x00, /* 14: Lch Input Mixer SW2 */ - 0x00, /* 15: Rch Input Mixer SW2 */ - 0x00, /* 16: Reset & Power Down */ - 0x00, /* 17: Clock Select */ - 0x00, /* 18: AD Input Select */ - 0x01 /* 19: Mic Amp Setup */ -}; - -int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, - struct snd_ak4531 **rak4531) -{ - unsigned int idx; - int err; - struct snd_ak4531 *ak4531; - static struct snd_device_ops ops = { - .dev_free = snd_ak4531_dev_free, - }; - - snd_assert(rak4531 != NULL, return -EINVAL); - *rak4531 = NULL; - snd_assert(card != NULL && _ak4531 != NULL, return -EINVAL); - ak4531 = kzalloc(sizeof(*ak4531), GFP_KERNEL); - if (ak4531 == NULL) - return -ENOMEM; - *ak4531 = *_ak4531; - mutex_init(&ak4531->reg_mutex); - if ((err = snd_component_add(card, "AK4531")) < 0) { - snd_ak4531_free(ak4531); - return err; - } - strcpy(card->mixername, "Asahi Kasei AK4531"); - ak4531->write(ak4531, AK4531_RESET, 0x03); /* no RST, PD */ - udelay(100); - ak4531->write(ak4531, AK4531_CLOCK, 0x00); /* CODEC ADC and CODEC DAC use {LR,B}CLK2 and run off LRCLK2 PLL */ - for (idx = 0; idx <= 0x19; idx++) { - if (idx == AK4531_RESET || idx == AK4531_CLOCK) - continue; - ak4531->write(ak4531, idx, ak4531->regs[idx] = snd_ak4531_initial_map[idx]); /* recording source is mixer */ - } - for (idx = 0; idx < ARRAY_SIZE(snd_ak4531_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ak4531_controls[idx], ak4531))) < 0) { - snd_ak4531_free(ak4531); - return err; - } - } - snd_ak4531_proc_init(card, ak4531); - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ak4531, &ops)) < 0) { - snd_ak4531_free(ak4531); - return err; - } - -#if 0 - snd_ak4531_dump(ak4531); -#endif - *rak4531 = ak4531; - return 0; -} - -/* - * power management - */ -#ifdef CONFIG_PM -void snd_ak4531_suspend(struct snd_ak4531 *ak4531) -{ - /* mute */ - ak4531->write(ak4531, AK4531_LMASTER, 0x9f); - ak4531->write(ak4531, AK4531_RMASTER, 0x9f); - /* powerdown */ - ak4531->write(ak4531, AK4531_RESET, 0x01); -} - -void snd_ak4531_resume(struct snd_ak4531 *ak4531) -{ - int idx; - - /* initialize */ - ak4531->write(ak4531, AK4531_RESET, 0x03); - udelay(100); - ak4531->write(ak4531, AK4531_CLOCK, 0x00); - /* restore mixer registers */ - for (idx = 0; idx <= 0x19; idx++) { - if (idx == AK4531_RESET || idx == AK4531_CLOCK) - continue; - ak4531->write(ak4531, idx, ak4531->regs[idx]); - } -} -#endif - -#ifdef CONFIG_PROC_FS -/* - * /proc interface - */ - -static void snd_ak4531_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct snd_ak4531 *ak4531 = entry->private_data; - - snd_iprintf(buffer, "Asahi Kasei AK4531\n\n"); - snd_iprintf(buffer, "Recording source : %s\n" - "MIC gain : %s\n", - ak4531->regs[AK4531_AD_IN] & 1 ? "external" : "mixer", - ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB"); -} - -static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "ak4531", &entry)) - snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); -} -#endif - -EXPORT_SYMBOL(snd_ak4531_mixer); -#ifdef CONFIG_PM -EXPORT_SYMBOL(snd_ak4531_suspend); -EXPORT_SYMBOL(snd_ak4531_resume); -#endif - -/* - * INIT part - */ - -static int __init alsa_ak4531_init(void) -{ - return 0; -} - -static void __exit alsa_ak4531_exit(void) -{ -} - -module_init(alsa_ak4531_init) -module_exit(alsa_ak4531_exit) diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c new file mode 100644 index 000000000000..6a99eed2d8b6 --- /dev/null +++ b/sound/pci/ak4531_codec.c @@ -0,0 +1,488 @@ +/* + * Copyright (c) by Jaroslav Kysela + * Universal routines for AK4531 codec + * + * + * 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 +#include +#include +#include + +#include +#include +#include + +MODULE_AUTHOR("Jaroslav Kysela "); +MODULE_DESCRIPTION("Universal routines for AK4531 codec"); +MODULE_LICENSE("GPL"); + +#ifdef CONFIG_PROC_FS +static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531); +#else +#define snd_ak4531_proc_init(card,ak) +#endif + +/* + * + */ + +#if 0 + +static void snd_ak4531_dump(struct snd_ak4531 *ak4531) +{ + int idx; + + for (idx = 0; idx < 0x19; idx++) + printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); +} + +#endif + +/* + * + */ + +#define AK4531_SINGLE(xname, xindex, reg, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_ak4531_info_single, \ + .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ + .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22) } +#define AK4531_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ak4531_info_single, \ + .get = snd_ak4531_get_single, .put = snd_ak4531_put_single, \ + .private_value = reg | (shift << 16) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + +static int snd_ak4531_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +static int snd_ak4531_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int val; + + mutex_lock(&ak4531->reg_mutex); + val = (ak4531->regs[reg] >> shift) & mask; + mutex_unlock(&ak4531->reg_mutex); + if (invert) { + val = mask - val; + } + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int change; + int val; + + val = ucontrol->value.integer.value[0] & mask; + if (invert) { + val = mask - val; + } + val <<= shift; + mutex_lock(&ak4531->reg_mutex); + val = (ak4531->regs[reg] & ~(mask << shift)) | val; + change = val != ak4531->regs[reg]; + ak4531->write(ak4531, reg, ak4531->regs[reg] = val); + mutex_unlock(&ak4531->reg_mutex); + return change; +} + +#define AK4531_DOUBLE(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_ak4531_info_double, \ + .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ + .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22) } +#define AK4531_DOUBLE_TLV(xname, xindex, left_reg, right_reg, left_shift, right_shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ak4531_info_double, \ + .get = snd_ak4531_get_double, .put = snd_ak4531_put_double, \ + .private_value = left_reg | (right_reg << 8) | (left_shift << 16) | (right_shift << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + +static int snd_ak4531_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +static int snd_ak4531_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int left_shift = (kcontrol->private_value >> 16) & 0x07; + int right_shift = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int left, right; + + mutex_lock(&ak4531->reg_mutex); + left = (ak4531->regs[left_reg] >> left_shift) & mask; + right = (ak4531->regs[right_reg] >> right_shift) & mask; + mutex_unlock(&ak4531->reg_mutex); + if (invert) { + left = mask - left; + right = mask - right; + } + ucontrol->value.integer.value[0] = left; + ucontrol->value.integer.value[1] = right; + return 0; +} + +static int snd_ak4531_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int left_shift = (kcontrol->private_value >> 16) & 0x07; + int right_shift = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int change; + int left, right; + + left = ucontrol->value.integer.value[0] & mask; + right = ucontrol->value.integer.value[1] & mask; + if (invert) { + left = mask - left; + right = mask - right; + } + left <<= left_shift; + right <<= right_shift; + mutex_lock(&ak4531->reg_mutex); + if (left_reg == right_reg) { + left = (ak4531->regs[left_reg] & ~((mask << left_shift) | (mask << right_shift))) | left | right; + change = left != ak4531->regs[left_reg]; + ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left); + } else { + left = (ak4531->regs[left_reg] & ~(mask << left_shift)) | left; + right = (ak4531->regs[right_reg] & ~(mask << right_shift)) | right; + change = left != ak4531->regs[left_reg] || right != ak4531->regs[right_reg]; + ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left); + ak4531->write(ak4531, right_reg, ak4531->regs[right_reg] = right); + } + mutex_unlock(&ak4531->reg_mutex); + return change; +} + +#define AK4531_INPUT_SW(xname, xindex, reg1, reg2, left_shift, right_shift) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_ak4531_info_input_sw, \ + .get = snd_ak4531_get_input_sw, .put = snd_ak4531_put_input_sw, \ + .private_value = reg1 | (reg2 << 8) | (left_shift << 16) | (right_shift << 24) } + +static int snd_ak4531_info_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 4; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_ak4531_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int reg1 = kcontrol->private_value & 0xff; + int reg2 = (kcontrol->private_value >> 8) & 0xff; + int left_shift = (kcontrol->private_value >> 16) & 0x0f; + int right_shift = (kcontrol->private_value >> 24) & 0x0f; + + mutex_lock(&ak4531->reg_mutex); + ucontrol->value.integer.value[0] = (ak4531->regs[reg1] >> left_shift) & 1; + ucontrol->value.integer.value[1] = (ak4531->regs[reg2] >> left_shift) & 1; + ucontrol->value.integer.value[2] = (ak4531->regs[reg1] >> right_shift) & 1; + ucontrol->value.integer.value[3] = (ak4531->regs[reg2] >> right_shift) & 1; + mutex_unlock(&ak4531->reg_mutex); + return 0; +} + +static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ak4531 *ak4531 = snd_kcontrol_chip(kcontrol); + int reg1 = kcontrol->private_value & 0xff; + int reg2 = (kcontrol->private_value >> 8) & 0xff; + int left_shift = (kcontrol->private_value >> 16) & 0x0f; + int right_shift = (kcontrol->private_value >> 24) & 0x0f; + int change; + int val1, val2; + + mutex_lock(&ak4531->reg_mutex); + val1 = ak4531->regs[reg1] & ~((1 << left_shift) | (1 << right_shift)); + val2 = ak4531->regs[reg2] & ~((1 << left_shift) | (1 << right_shift)); + val1 |= (ucontrol->value.integer.value[0] & 1) << left_shift; + val2 |= (ucontrol->value.integer.value[1] & 1) << left_shift; + val1 |= (ucontrol->value.integer.value[2] & 1) << right_shift; + val2 |= (ucontrol->value.integer.value[3] & 1) << right_shift; + change = val1 != ak4531->regs[reg1] || val2 != ak4531->regs[reg2]; + ak4531->write(ak4531, reg1, ak4531->regs[reg1] = val1); + ak4531->write(ak4531, reg2, ak4531->regs[reg2] = val2); + mutex_unlock(&ak4531->reg_mutex); + return change; +} + +static const DECLARE_TLV_DB_SCALE(db_scale_master, -6200, 200, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_mono, -2800, 400, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_input, -5000, 200, 0); + +static struct snd_kcontrol_new snd_ak4531_controls[] __devinitdata = { + +AK4531_DOUBLE_TLV("Master Playback Switch", 0, + AK4531_LMASTER, AK4531_RMASTER, 7, 7, 1, 1, + db_scale_master), +AK4531_DOUBLE("Master Playback Volume", 0, AK4531_LMASTER, AK4531_RMASTER, 0, 0, 0x1f, 1), + +AK4531_SINGLE_TLV("Master Mono Playback Switch", 0, AK4531_MONO_OUT, 7, 1, 1, + db_scale_mono), +AK4531_SINGLE("Master Mono Playback Volume", 0, AK4531_MONO_OUT, 0, 0x07, 1), + +AK4531_DOUBLE("PCM Switch", 0, AK4531_LVOICE, AK4531_RVOICE, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("PCM Volume", 0, AK4531_LVOICE, AK4531_RVOICE, 0, 0, 0x1f, 1, + db_scale_input), +AK4531_DOUBLE("PCM Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 3, 2, 1, 0), +AK4531_DOUBLE("PCM Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 2, 2, 1, 0), + +AK4531_DOUBLE("PCM Switch", 1, AK4531_LFM, AK4531_RFM, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("PCM Volume", 1, AK4531_LFM, AK4531_RFM, 0, 0, 0x1f, 1, + db_scale_input), +AK4531_DOUBLE("PCM Playback Switch", 1, AK4531_OUT_SW1, AK4531_OUT_SW1, 6, 5, 1, 0), +AK4531_INPUT_SW("PCM Capture Route", 1, AK4531_LIN_SW1, AK4531_RIN_SW1, 6, 5), + +AK4531_DOUBLE("CD Switch", 0, AK4531_LCD, AK4531_RCD, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("CD Volume", 0, AK4531_LCD, AK4531_RCD, 0, 0, 0x1f, 1, + db_scale_input), +AK4531_DOUBLE("CD Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 2, 1, 1, 0), +AK4531_INPUT_SW("CD Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 2, 1), + +AK4531_DOUBLE("Line Switch", 0, AK4531_LLINE, AK4531_RLINE, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("Line Volume", 0, AK4531_LLINE, AK4531_RLINE, 0, 0, 0x1f, 1, + db_scale_input), +AK4531_DOUBLE("Line Playback Switch", 0, AK4531_OUT_SW1, AK4531_OUT_SW1, 4, 3, 1, 0), +AK4531_INPUT_SW("Line Capture Route", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 4, 3), + +AK4531_DOUBLE("Aux Switch", 0, AK4531_LAUXA, AK4531_RAUXA, 7, 7, 1, 1), +AK4531_DOUBLE_TLV("Aux Volume", 0, AK4531_LAUXA, AK4531_RAUXA, 0, 0, 0x1f, 1, + db_scale_input), +AK4531_DOUBLE("Aux Playback Switch", 0, AK4531_OUT_SW2, AK4531_OUT_SW2, 5, 4, 1, 0), +AK4531_INPUT_SW("Aux Capture Route", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 4, 3), + +AK4531_SINGLE("Mono Switch", 0, AK4531_MONO1, 7, 1, 1), +AK4531_SINGLE_TLV("Mono Volume", 0, AK4531_MONO1, 0, 0x1f, 1, db_scale_input), +AK4531_SINGLE("Mono Playback Switch", 0, AK4531_OUT_SW2, 0, 1, 0), +AK4531_DOUBLE("Mono Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 0, 0, 1, 0), + +AK4531_SINGLE("Mono Switch", 1, AK4531_MONO2, 7, 1, 1), +AK4531_SINGLE_TLV("Mono Volume", 1, AK4531_MONO2, 0, 0x1f, 1, db_scale_input), +AK4531_SINGLE("Mono Playback Switch", 1, AK4531_OUT_SW2, 1, 1, 0), +AK4531_DOUBLE("Mono Capture Switch", 1, AK4531_LIN_SW2, AK4531_RIN_SW2, 1, 1, 1, 0), + +AK4531_SINGLE_TLV("Mic Volume", 0, AK4531_MIC, 0, 0x1f, 1, db_scale_input), +AK4531_SINGLE("Mic Switch", 0, AK4531_MIC, 7, 1, 1), +AK4531_SINGLE("Mic Playback Switch", 0, AK4531_OUT_SW1, 0, 1, 0), +AK4531_DOUBLE("Mic Capture Switch", 0, AK4531_LIN_SW1, AK4531_RIN_SW1, 0, 0, 1, 0), + +AK4531_DOUBLE("Mic Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 7, 7, 1, 0), +AK4531_DOUBLE("Mono1 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 6, 6, 1, 0), +AK4531_DOUBLE("Mono2 Bypass Capture Switch", 0, AK4531_LIN_SW2, AK4531_RIN_SW2, 5, 5, 1, 0), + +AK4531_SINGLE("AD Input Select", 0, AK4531_AD_IN, 0, 1, 0), +AK4531_SINGLE("Mic Boost (+30dB)", 0, AK4531_MIC_GAIN, 0, 1, 0) +}; + +static int snd_ak4531_free(struct snd_ak4531 *ak4531) +{ + if (ak4531) { + if (ak4531->private_free) + ak4531->private_free(ak4531); + kfree(ak4531); + } + return 0; +} + +static int snd_ak4531_dev_free(struct snd_device *device) +{ + struct snd_ak4531 *ak4531 = device->device_data; + return snd_ak4531_free(ak4531); +} + +static u8 snd_ak4531_initial_map[0x19 + 1] = { + 0x9f, /* 00: Master Volume Lch */ + 0x9f, /* 01: Master Volume Rch */ + 0x9f, /* 02: Voice Volume Lch */ + 0x9f, /* 03: Voice Volume Rch */ + 0x9f, /* 04: FM Volume Lch */ + 0x9f, /* 05: FM Volume Rch */ + 0x9f, /* 06: CD Audio Volume Lch */ + 0x9f, /* 07: CD Audio Volume Rch */ + 0x9f, /* 08: Line Volume Lch */ + 0x9f, /* 09: Line Volume Rch */ + 0x9f, /* 0a: Aux Volume Lch */ + 0x9f, /* 0b: Aux Volume Rch */ + 0x9f, /* 0c: Mono1 Volume */ + 0x9f, /* 0d: Mono2 Volume */ + 0x9f, /* 0e: Mic Volume */ + 0x87, /* 0f: Mono-out Volume */ + 0x00, /* 10: Output Mixer SW1 */ + 0x00, /* 11: Output Mixer SW2 */ + 0x00, /* 12: Lch Input Mixer SW1 */ + 0x00, /* 13: Rch Input Mixer SW1 */ + 0x00, /* 14: Lch Input Mixer SW2 */ + 0x00, /* 15: Rch Input Mixer SW2 */ + 0x00, /* 16: Reset & Power Down */ + 0x00, /* 17: Clock Select */ + 0x00, /* 18: AD Input Select */ + 0x01 /* 19: Mic Amp Setup */ +}; + +int __devinit snd_ak4531_mixer(struct snd_card *card, + struct snd_ak4531 *_ak4531, + struct snd_ak4531 **rak4531) +{ + unsigned int idx; + int err; + struct snd_ak4531 *ak4531; + static struct snd_device_ops ops = { + .dev_free = snd_ak4531_dev_free, + }; + + snd_assert(rak4531 != NULL, return -EINVAL); + *rak4531 = NULL; + snd_assert(card != NULL && _ak4531 != NULL, return -EINVAL); + ak4531 = kzalloc(sizeof(*ak4531), GFP_KERNEL); + if (ak4531 == NULL) + return -ENOMEM; + *ak4531 = *_ak4531; + mutex_init(&ak4531->reg_mutex); + if ((err = snd_component_add(card, "AK4531")) < 0) { + snd_ak4531_free(ak4531); + return err; + } + strcpy(card->mixername, "Asahi Kasei AK4531"); + ak4531->write(ak4531, AK4531_RESET, 0x03); /* no RST, PD */ + udelay(100); + ak4531->write(ak4531, AK4531_CLOCK, 0x00); /* CODEC ADC and CODEC DAC use {LR,B}CLK2 and run off LRCLK2 PLL */ + for (idx = 0; idx <= 0x19; idx++) { + if (idx == AK4531_RESET || idx == AK4531_CLOCK) + continue; + ak4531->write(ak4531, idx, ak4531->regs[idx] = snd_ak4531_initial_map[idx]); /* recording source is mixer */ + } + for (idx = 0; idx < ARRAY_SIZE(snd_ak4531_controls); idx++) { + if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_ak4531_controls[idx], ak4531))) < 0) { + snd_ak4531_free(ak4531); + return err; + } + } + snd_ak4531_proc_init(card, ak4531); + if ((err = snd_device_new(card, SNDRV_DEV_CODEC, ak4531, &ops)) < 0) { + snd_ak4531_free(ak4531); + return err; + } + +#if 0 + snd_ak4531_dump(ak4531); +#endif + *rak4531 = ak4531; + return 0; +} + +/* + * power management + */ +#ifdef CONFIG_PM +void snd_ak4531_suspend(struct snd_ak4531 *ak4531) +{ + /* mute */ + ak4531->write(ak4531, AK4531_LMASTER, 0x9f); + ak4531->write(ak4531, AK4531_RMASTER, 0x9f); + /* powerdown */ + ak4531->write(ak4531, AK4531_RESET, 0x01); +} + +void snd_ak4531_resume(struct snd_ak4531 *ak4531) +{ + int idx; + + /* initialize */ + ak4531->write(ak4531, AK4531_RESET, 0x03); + udelay(100); + ak4531->write(ak4531, AK4531_CLOCK, 0x00); + /* restore mixer registers */ + for (idx = 0; idx <= 0x19; idx++) { + if (idx == AK4531_RESET || idx == AK4531_CLOCK) + continue; + ak4531->write(ak4531, idx, ak4531->regs[idx]); + } +} +#endif + +#ifdef CONFIG_PROC_FS +/* + * /proc interface + */ + +static void snd_ak4531_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ak4531 *ak4531 = entry->private_data; + + snd_iprintf(buffer, "Asahi Kasei AK4531\n\n"); + snd_iprintf(buffer, "Recording source : %s\n" + "MIC gain : %s\n", + ak4531->regs[AK4531_AD_IN] & 1 ? "external" : "mixer", + ak4531->regs[AK4531_MIC_GAIN] & 1 ? "+30dB" : "+0dB"); +} + +static void __devinit +snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531) +{ + struct snd_info_entry *entry; + + if (! snd_card_proc_new(card, "ak4531", &entry)) + snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read); +} +#endif -- cgit v1.2.3 From 8bb8b453cb458d8f62411e78a4cfd6d860b503b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:47:45 +0200 Subject: [ALSA] trident - clean up obsolete synth codes Clean up the unused synth codes in the memory handling of trident driver. Signed-off-by: Takashi Iwai --- sound/pci/trident/trident_memory.c | 178 ------------------------------------- 1 file changed, 178 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index df9b487fa17e..3fd7f1b29b0f 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -310,181 +310,3 @@ int snd_trident_free_pages(struct snd_trident *trident, mutex_unlock(&hdr->block_mutex); return 0; } - - -/*---------------------------------------------------------------- - * memory allocation using multiple pages (for synth) - *---------------------------------------------------------------- - * Unlike the DMA allocation above, non-contiguous pages are - * assigned to TLB. - *----------------------------------------------------------------*/ - -/* - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk); -static int synth_free_pages(struct snd_trident *hw, struct snd_util_memblk *blk); - -/* - * allocate a synth sample area - */ -struct snd_util_memblk * -snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) -{ - struct snd_util_memblk *blk; - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - blk = __snd_util_mem_alloc(hdr, size); - if (blk == NULL) { - mutex_unlock(&hdr->block_mutex); - return NULL; - } - if (synth_alloc_pages(hw, blk)) { - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return NULL; - } - mutex_unlock(&hdr->block_mutex); - return blk; -} - -EXPORT_SYMBOL(snd_trident_synth_alloc); - -/* - * free a synth sample area - */ -int -snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - struct snd_util_memhdr *hdr = hw->tlb.memhdr; - - mutex_lock(&hdr->block_mutex); - synth_free_pages(hw, blk); - __snd_util_mem_free(hdr, blk); - mutex_unlock(&hdr->block_mutex); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_free); - -/* - * reset TLB entry and free kernel page - */ -static void clear_tlb(struct snd_trident *trident, int page) -{ - void *ptr = page_to_ptr(trident, page); - dma_addr_t addr = page_to_addr(trident, page); - set_silent_tlb(trident, page); - if (ptr) { - struct snd_dma_buffer dmab; - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(trident->pci); - dmab.area = ptr; - dmab.addr = addr; - dmab.bytes = ALIGN_PAGE_SIZE; - snd_dma_free_pages(&dmab); - } -} - -/* check new allocation range */ -static void get_single_page_range(struct snd_util_memhdr *hdr, - struct snd_util_memblk *blk, - int *first_page_ret, int *last_page_ret) -{ - struct list_head *p; - struct snd_util_memblk *q; - int first_page, last_page; - first_page = firstpg(blk); - if ((p = blk->list.prev) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (lastpg(q) == first_page) - first_page++; /* first page was already allocated */ - } - last_page = lastpg(blk); - if ((p = blk->list.next) != &hdr->block) { - q = list_entry(p, struct snd_util_memblk, list); - if (firstpg(q) == last_page) - last_page--; /* last page was already allocated */ - } - *first_page_ret = first_page; - *last_page_ret = last_page; -} - -/* - * allocate kernel pages and assign them to TLB - */ -static int synth_alloc_pages(struct snd_trident *hw, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - struct snd_dma_buffer dmab; - - firstpg(blk) = get_aligned_page(blk->offset); - lastpg(blk) = get_aligned_page(blk->offset + blk->size - 1); - get_single_page_range(hw->tlb.memhdr, blk, &first_page, &last_page); - - /* allocate a kernel page for each Trident page - - * fortunately Trident page size and kernel PAGE_SIZE is identical! - */ - for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(hw->pci), - ALIGN_PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; - } - set_tlb_bus(hw, page, (unsigned long)dmab.area, dmab.addr); - } - return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) - clear_tlb(hw, page); - - return -ENOMEM; -} - -/* - * free pages - */ -static int synth_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk) -{ - int page, first_page, last_page; - - get_single_page_range(trident->tlb.memhdr, blk, &first_page, &last_page); - for (page = first_page; page <= last_page; page++) - clear_tlb(trident, page); - - return 0; -} - -/* - * copy_from_user(blk + offset, data, size) - */ -int snd_trident_synth_copy_from_user(struct snd_trident *trident, - struct snd_util_memblk *blk, - int offset, const char __user *data, int size) -{ - int page, nextofs, end_offset, temp, temp1; - - offset += blk->offset; - end_offset = offset + size; - page = get_aligned_page(offset) + 1; - do { - nextofs = aligned_page_offset(page); - temp = nextofs - offset; - temp1 = end_offset - offset; - if (temp1 < temp) - temp = temp1; - if (copy_from_user(offset_ptr(trident, offset), data, temp)) - return -EFAULT; - offset = nextofs; - data += temp; - page++; - } while (offset < end_offset); - return 0; -} - -EXPORT_SYMBOL(snd_trident_synth_copy_from_user); -- cgit v1.2.3 From a5003fc04113c217370409beac812831cbf6e0ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 May 2008 09:49:41 +0200 Subject: [ALSA] emu10k1 - simplify page allocation for synth Simplify the page allocation of emu10k1 driver for emux synth support. Since these pages aren't be necessarily coherent, we can avoid expensive DMA-coherent routines. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 66 ++++++++++++++++++++-------------------------- 1 file changed, 28 insertions(+), 38 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 916c1dbcd53c..128eaca17a61 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -437,43 +437,46 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, *last_page_ret = last_page; } +/* release allocated pages */ +static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, + int last_page) +{ + int page; + + for (page = first_page; page <= last_page; page++) { + free_page((unsigned long)emu->page_ptr_table[page]); + emu->page_addr_table[page] = 0; + emu->page_ptr_table[page] = NULL; + } +} + /* * allocate kernel pages */ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; - struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) - goto __fail; - if (! is_valid_page(emu, dmab.addr)) { - snd_dma_free_pages(&dmab); - goto __fail; + /* first try to allocate from <4GB zone */ + struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | + __GFP_NOWARN); + if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) + /* try to allocate from <16MB zone */ + p = alloc_page(GFP_DMA | + __GFP_NORETRY | /* no OOM-killer */ + __GFP_NOWARN); + if (!p) { + __synth_free_pages(emu, first_page, page - 1); + return -ENOMEM; } - emu->page_addr_table[page] = dmab.addr; - emu->page_ptr_table[page] = dmab.area; + emu->page_addr_table[page] = page_to_phys(p); + emu->page_ptr_table[page] = page_address(p); } return 0; - -__fail: - /* release allocated pages */ - last_page = page - 1; - for (page = first_page; page <= last_page; page++) { - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - - return -ENOMEM; } /* @@ -481,23 +484,10 @@ __fail: */ static int synth_free_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { - int page, first_page, last_page; - struct snd_dma_buffer dmab; + int first_page, last_page; get_single_page_range(emu->memhdr, blk, &first_page, &last_page); - dmab.dev.type = SNDRV_DMA_TYPE_DEV; - dmab.dev.dev = snd_dma_pci_data(emu->pci); - for (page = first_page; page <= last_page; page++) { - if (emu->page_ptr_table[page] == NULL) - continue; - dmab.area = emu->page_ptr_table[page]; - dmab.addr = emu->page_addr_table[page]; - dmab.bytes = PAGE_SIZE; - snd_dma_free_pages(&dmab); - emu->page_addr_table[page] = 0; - emu->page_ptr_table[page] = NULL; - } - + __synth_free_pages(emu, first_page, last_page); return 0; } -- cgit v1.2.3 From d2cd74b158d7214a556226e3312f9fb1de64d7ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Jun 2008 11:45:53 +0200 Subject: [ALSA] emu10k1 - Fix inverted Analog/Digital mixer switch on Audigy2 On Audigy2 Platinum, the Analog/Digital mixer switch is inverted. https://bugzilla.novell.com/show_bug.cgi?id=396204 The patch adds a simple workaround. There might be another device requiring a similar fix, too (or fix for audigy2 generically), but right now I fix only the known broken one. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 1 + sound/pci/emu10k1/emumixer.c | 13 ++++++++++--- 2 files changed, 11 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 548c9cc81af5..2f283ea6ad9a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index fd221209abcb..f34bbfb705f5 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1578,6 +1578,10 @@ static int snd_emu10k1_shared_spdif_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = inl(emu->port + A_IOCFG) & A_IOCFG_GPOUT0 ? 1 : 0; else ucontrol->value.integer.value[0] = inl(emu->port + HCFG) & HCFG_GPOUT0 ? 1 : 0; + if (emu->card_capabilities->invert_shared_spdif) + ucontrol->value.integer.value[0] = + !ucontrol->value.integer.value[0]; + return 0; } @@ -1586,15 +1590,18 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, { unsigned long flags; struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); - unsigned int reg, val; + unsigned int reg, val, sw; int change = 0; + sw = ucontrol->value.integer.value[0]; + if (emu->card_capabilities->invert_shared_spdif) + sw = !sw; spin_lock_irqsave(&emu->reg_lock, flags); if ( emu->card_capabilities->i2c_adc) { /* Do nothing for Audigy 2 ZS Notebook */ } else if (emu->audigy) { reg = inl(emu->port + A_IOCFG); - val = ucontrol->value.integer.value[0] ? A_IOCFG_GPOUT0 : 0; + val = sw ? A_IOCFG_GPOUT0 : 0; change = (reg & A_IOCFG_GPOUT0) != val; if (change) { reg &= ~A_IOCFG_GPOUT0; @@ -1603,7 +1610,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol, } } reg = inl(emu->port + HCFG); - val = ucontrol->value.integer.value[0] ? HCFG_GPOUT0 : 0; + val = sw ? HCFG_GPOUT0 : 0; change |= (reg & HCFG_GPOUT0) != val; if (change) { reg &= ~HCFG_GPOUT0; -- cgit v1.2.3 From 17bba1b72d190742c99a140154f0abac9c1996c3 Mon Sep 17 00:00:00 2001 From: Jiang zhe Date: Wed, 4 Jun 2008 12:11:07 +0200 Subject: [ALSA] hda - support intel DG33 motherboards These two motherboards's pin configuration are not covered by driver. I wrote a new model to support them. Signed-off-by: Jiang zhe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 107 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 107 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cb3e0283f0ce..048cb84721fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -209,6 +209,7 @@ enum { ALC883_MITAC, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, + ALC883_3ST_6ch_INTEL, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6603,6 +6604,16 @@ static struct hda_input_mux alc883_capture_source = { }, }; +static struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_lenovo_101e_capture_source = { .num_items = 2, .items = { @@ -6683,6 +6694,48 @@ static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { { 6, alc883_3ST_ch6_init }, }; +/* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { + { 2, alc883_3ST_ch2_intel_init }, + { 4, alc883_3ST_ch4_intel_init }, + { 6, alc883_3ST_ch6_intel_init }, +}; + /* * 6ch mode */ @@ -6915,6 +6968,45 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -7763,6 +7855,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_MITAC] = "mitac", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", + [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC883_AUTO] = "auto", }; @@ -7820,6 +7913,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; @@ -7858,6 +7953,18 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_3ST_6ch_INTEL] = { + .mixers = { alc883_3ST_6ch_intel_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), + .channel_mode = alc883_3ST_6ch_intel_modes, + .need_dac_fix = 1, + .input_mux = &alc883_3stack_6ch_intel, + }, [ALC883_6ST_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs }, -- cgit v1.2.3 From 2c3bf9abb11dd8050cd2d153917d1746c8d5af05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jun 2008 12:39:38 +0200 Subject: [ALSA] hda - Fix PLL gating control on Realtek codecs On some Realtek codecs, the analog PLL gating control bit must be set off while the default value is 1. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 048cb84721fa..7997e13b59a6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -285,6 +285,10 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif + + /* for PLL fix */ + hda_nid_t pll_nid; + unsigned int pll_coef_idx, pll_coef_bit; }; /* @@ -752,6 +756,38 @@ static struct hda_verb alc_gpio3_init_verbs[] = { { } }; +/* + * Fix hardware PLL issue + * On some codecs, the analog PLL gating control must be off while + * the default value is 1. + */ +static void alc_fix_pll(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (!spec->pll_nid) + return; + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + val = snd_hda_codec_read(codec, spec->pll_nid, 0, + AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, + spec->pll_coef_idx); + snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, + val & ~(1 << spec->pll_coef_bit)); +} + +static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_bit) +{ + struct alc_spec *spec = codec->spec; + spec->pll_nid = nid; + spec->pll_coef_idx = coef_idx; + spec->pll_coef_bit = coef_bit; + alc_fix_pll(codec); +} + static void alc_sku_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2400,6 +2436,8 @@ static int alc_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; unsigned int i; + alc_fix_pll(codec); + for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -8286,6 +8324,8 @@ static int patch_alc883(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); @@ -9886,6 +9926,8 @@ static int patch_alc262(struct hda_codec *codec) } #endif + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + board_config = snd_hda_check_board_config(codec, ALC262_MODEL_LAST, alc262_models, alc262_cfg_tbl); @@ -11196,6 +11238,8 @@ static int patch_alc269(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -14531,6 +14575,8 @@ static int patch_alc662(struct hda_codec *codec) codec->spec = spec; + alc_fix_pll_init(codec, 0x20, 0x04, 15); + board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, alc662_cfg_tbl); -- cgit v1.2.3 From 607d982bbea2a14b5b77cc7689f509d588e1e6da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jun 2008 12:41:21 +0200 Subject: [ALSA] hda - increase max_codecs of ICH to 4 It turned out that some ICH9-based boards use SD3 for the audio codec where the current driver code doesn't probe. Since we have a better codec slot check now, it must be safe to increase this to 4. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e2dc4f5ca7a..dc68709e7569 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1104,7 +1104,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) */ static unsigned int azx_max_codecs[] __devinitdata = { - [AZX_DRIVER_ICH] = 3, + [AZX_DRIVER_ICH] = 4, /* Some ICH9 boards use SD3 */ [AZX_DRIVER_SCH] = 3, [AZX_DRIVER_ATI] = 4, [AZX_DRIVER_ATIHDMI] = 4, -- cgit v1.2.3 From 8b83afe0d21f07145ec34ac006656e4d3edc4bac Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Ville=20Syrj=C3=A4l=C3=A4?= Date: Tue, 3 Jun 2008 20:52:10 +0300 Subject: [ALSA] maestro3: Fix hw volume on HP OmniBook Make the hw volume buttons work correctly on some HP OmniBook laptops. The original quirk was apparently applied a bit too early and it was also lacking some critial register writes. This improved sequence was discovered by trial and error (like the original sequence). Tested and found working on OB500 and OB6000 laptops. Signed-off-by: Ville Syrjala Signed-off-by: Takashi Iwai --- sound/pci/maestro3.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index a536c59fbea1..f4788dee05c3 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, int enable) outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2642,6 +2650,8 @@ static int m3_resume(struct pci_dev *pci) snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2781,6 +2791,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, -- cgit v1.2.3 From 0264b3b636f1ba5777426dd9b5bb5cbbeb5fcb34 Mon Sep 17 00:00:00 2001 From: Juergen Beisert Date: Fri, 6 Jun 2008 17:02:57 +0200 Subject: [ALSA] ac97: add support for wm9711 master left inv switch This patch adds support for Master Left Inv Switch on wm9711. At least required to drive the mono speaker on the PXA270 platfrom Signed-off-by: Juergen Beisert Acked-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 92817f7d46d2..abe88adec37e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -669,6 +669,7 @@ AC97_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), AC97_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), AC97_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), +AC97_SINGLE("Master Left Inv Switch", AC97_MASTER, 6, 1, 0), AC97_SINGLE("Master ZC Switch", AC97_MASTER, 7, 1, 0), AC97_SINGLE("Headphone ZC Switch", AC97_HEADPHONE, 7, 1, 0), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -- cgit v1.2.3 From 38f582270feb19ea8821cb64c5d2fdbd6bdd0bd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 8 Jun 2008 09:17:27 +0200 Subject: [ALSA] ac97 - Fix power_save option value as time-out The power_save option was set as boot although it was meant to be a timeout value like the same option of snd-hda-intel originally. Now fixed to the same style. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 45fd29017ddd..2d2f16e11082 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -49,8 +49,9 @@ MODULE_PARM_DESC(enable_loopback, "Enable AC97 ADC/DAC Loopback Control"); #ifdef CONFIG_SND_AC97_POWER_SAVE static int power_save = CONFIG_SND_AC97_POWER_SAVE_DEFAULT; -module_param(power_save, bool, 0644); -MODULE_PARM_DESC(power_save, "Enable AC97 power-saving control"); +module_param(power_save, int, 0644); +MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " + "(in second, 0 = disable)."); #endif /* @@ -2362,7 +2363,7 @@ int snd_ac97_update_power(struct snd_ac97 *ac97, int reg, int powerup) * that open/close frequently) */ schedule_delayed_work(&ac97->power_work, - msecs_to_jiffies(2000)); + msecs_to_jiffies(power_save * 1000)); else { cancel_delayed_work(&ac97->power_work); update_power_regs(ac97); -- cgit v1.2.3 From 36ca6e133ae3a57318a3ad8a41b2f005e3934c29 Mon Sep 17 00:00:00 2001 From: Akio Idehara Date: Mon, 9 Jun 2008 22:57:40 +0900 Subject: [ALSA] hda - Add Toshiba dynabook SS RX1 support I have Toshiba dynabook SS RX1 and this patch adds that support. Signed-off-by: Akio Idehara Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7997e13b59a6..61f8c13b2bb9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9719,6 +9719,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), -- cgit v1.2.3 From 781711a93e8aae756e8ae07acef185a6dfe552b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 12:34:45 +0200 Subject: ALSA: emu10k1 - Fix page allocation with GFP_DMA Added the missing GFP_ATOMIC to page_alloc when called with GFP_DMA. GFP_KERNEL often results in stalls for ZONE_DMA, so GFP_ATOMIC is more prgmatic. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 128eaca17a61..42943b4fcb7b 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -466,7 +466,7 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk __GFP_NOWARN); if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) /* try to allocate from <16MB zone */ - p = alloc_page(GFP_DMA | + p = alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); if (!p) { -- cgit v1.2.3 From 675f25d4d3ff6501cbce608bcc2333a56ec4c105 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:20 +0200 Subject: ALSA: hda - Add bdl_pos_adj option Added a new option, bdl_pos_adj, to adjust the delay of IRQ-wakeup timing. Most HD-audio hardwares have a problem that a BDL IRQ is issued before actually the data and the DMA pointer are updated. We have already a mechanism to force to delay snd_pcm_period_elapsed() calls via workq, but this costs much CPU, and typically the delay is within one sample. Thus, it's more clever to adjust the BDL entries instead. The new option adds the size of the delay in frames. As default, it's set to 1 -- that is, one sample delay. Even the hardware is really correct, one sample delay is relatively harmless in comparison with reporting wrong positions. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 117 +++++++++++++++++++++++++++++++++------------- 1 file changed, 85 insertions(+), 32 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dc68709e7569..1ec3fd4c8940 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -58,6 +58,7 @@ static int position_fix[SNDRV_CARDS]; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; +static int bdl_pos_adj = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -77,6 +78,8 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); +module_param(bdl_pos_adj, int, 0644); +MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset"); #ifdef CONFIG_SND_HDA_POWER_SAVE /* power_save option is defined in hda_codec.c */ @@ -309,7 +312,8 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; - unsigned int irq_pending: 1; + unsigned int irq_pending :1; + unsigned int irq_ignore :1; }; /* CORB/RIRB */ @@ -943,6 +947,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); if (!azx_dev->substream || !azx_dev->running) continue; + /* ignore the first dummy IRQ (due to pos_adj) */ + if (azx_dev->irq_ignore) { + azx_dev->irq_ignore = 0; + continue; + } /* check whether this IRQ is really acceptable */ if (azx_position_ok(chip, azx_dev)) { azx_dev->irq_pending = 0; @@ -976,15 +985,54 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) } +/* + * set up a BDL entry + */ +static int setup_bdle(struct snd_pcm_substream *substream, + struct azx_dev *azx_dev, u32 **bdlp, + int ofs, int size, int with_ioc) +{ + struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); + u32 *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES) + return -EINVAL; + + addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32bit(addr)); + /* program the size field of the BDL entry */ + chunk = PAGE_SIZE - (ofs % PAGE_SIZE); + if (size < chunk) + chunk = size; + bdl[2] = cpu_to_le32(chunk); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + size -= chunk; + bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); + bdl += 4; + azx_dev->frags++; + ofs += chunk; + } + *bdlp = bdl; + return ofs; +} + /* * set up BDL entries */ static int azx_setup_periods(struct snd_pcm_substream *substream, struct azx_dev *azx_dev) { - struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); u32 *bdl; int i, ofs, periods, period_bytes; + int pos_adj = 0; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); @@ -998,39 +1046,44 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, bdl = (u32 *)azx_dev->bdl.area; ofs = 0; azx_dev->frags = 0; - for (i = 0; i < periods; i++) { - int size, rest; - if (i >= AZX_MAX_BDL_ENTRIES) { - snd_printk(KERN_ERR "Too many BDL entries: " - "buffer=%d, period=%d\n", - azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - return -EINVAL; + azx_dev->irq_ignore = 0; + if (bdl_pos_adj > 0) { + struct snd_pcm_runtime *runtime = substream->runtime; + pos_adj = (bdl_pos_adj * runtime->rate + 47999) / 48000; + if (!pos_adj) + pos_adj = 1; + pos_adj = frames_to_bytes(runtime, pos_adj); + if (pos_adj >= period_bytes) { + snd_printk(KERN_WARNING "Too big adjustment %d\n", + bdl_pos_adj); + pos_adj = 0; + } else { + ofs = setup_bdle(substream, azx_dev, + &bdl, ofs, pos_adj, 1); + if (ofs < 0) + goto error; + azx_dev->irq_ignore = 1; } - rest = period_bytes; - do { - dma_addr_t addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); - /* program the address field of the BDL entry */ - bdl[0] = cpu_to_le32((u32)addr); - bdl[1] = cpu_to_le32(upper_32bit(addr)); - /* program the size field of the BDL entry */ - size = PAGE_SIZE - (ofs % PAGE_SIZE); - if (rest < size) - size = rest; - bdl[2] = cpu_to_le32(size); - /* program the IOC to enable interrupt - * only when the whole fragment is processed - */ - rest -= size; - bdl[3] = rest ? 0 : cpu_to_le32(0x01); - bdl += 4; - azx_dev->frags++; - ofs += size; - } while (rest > 0); + } + for (i = 0; i < periods; i++) { + if (i == periods - 1 && pos_adj) + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes - pos_adj, 0); + else + ofs = setup_bdle(substream, azx_dev, &bdl, ofs, + period_bytes, 1); + if (ofs < 0) + goto error; } return 0; + + error: + snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", + azx_dev->bufsize, period_bytes); + /* reset */ + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + return -EINVAL; } /* -- cgit v1.2.3 From d2e1c97331d37a477ef7e9a272b4de4e53b893be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:34 +0200 Subject: ALSA: hda - remove position_fix=3 position_fix=3 is the option to correct the DMA position with the FIFO size. But, it never worked correctly, and we have now more other workarounds for the DMA position fixes. Thus better to remove it. Also, change POS_FIX_NONE to POS_FIX_LPIB to represent its real role better. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ec3fd4c8940..0ff8b9b9629e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -70,7 +70,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " - "(0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); + "(0 = auto, 1 = none, 2 = POSBUF)."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); @@ -266,9 +266,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* position fix mode */ enum { POS_FIX_AUTO, - POS_FIX_NONE, + POS_FIX_LPIB, POS_FIX_POSBUF, - POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -1506,8 +1505,6 @@ static unsigned int azx_get_position(struct azx *chip, } else { /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); - if (chip->position_fix == POS_FIX_FIFO) - pos += azx_dev->fifo_size; } if (pos >= azx_dev->bufsize) pos = 0; @@ -1542,7 +1539,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) printk(KERN_WARNING "hda-intel: Invalid position buffer, " "using LPIB read method instead.\n"); - chip->position_fix = POS_FIX_NONE; + chip->position_fix = POS_FIX_LPIB; pos = azx_get_position(chip, azx_dev); } else chip->position_fix = POS_FIX_POSBUF; @@ -1917,9 +1914,9 @@ static int azx_dev_free(struct snd_device *device) * white/black-listing for position_fix */ static struct snd_pci_quirk position_fix_list[] __devinitdata = { - SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_NONE), - SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_NONE), - SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_NONE), + SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), {} }; -- cgit v1.2.3 From 555e219ffaeecbd08d1603e1551188483e7e3b64 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:34 +0200 Subject: ALSA: hda - bdl_pos_adj option to each instance The option bdl_pos_adj should be provided for each card instance instead of a global one because the value depends rather on each controller-chip. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 25 +++++++++++++++---------- 1 file changed, 15 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0ff8b9b9629e..ddae3c479a88 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,10 +55,10 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; +static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; -static int bdl_pos_adj = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -71,6 +71,8 @@ MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer " "(0 = auto, 1 = none, 2 = POSBUF)."); +module_param_array(bdl_pos_adj, int, NULL, 0644); +MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); module_param(single_cmd, bool, 0444); @@ -78,8 +80,6 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); -module_param(bdl_pos_adj, int, 0644); -MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset"); #ifdef CONFIG_SND_HDA_POWER_SAVE /* power_save option is defined in hda_codec.c */ @@ -330,6 +330,7 @@ struct azx_rb { struct azx { struct snd_card *card; struct pci_dev *pci; + int dev_index; /* chip type specific */ int driver_type; @@ -1026,12 +1027,13 @@ static int setup_bdle(struct snd_pcm_substream *substream, /* * set up BDL entries */ -static int azx_setup_periods(struct snd_pcm_substream *substream, +static int azx_setup_periods(struct azx *chip, + struct snd_pcm_substream *substream, struct azx_dev *azx_dev) { u32 *bdl; int i, ofs, periods, period_bytes; - int pos_adj = 0; + int pos_adj; /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); @@ -1046,15 +1048,16 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, ofs = 0; azx_dev->frags = 0; azx_dev->irq_ignore = 0; - if (bdl_pos_adj > 0) { + pos_adj = bdl_pos_adj[chip->dev_index]; + if (pos_adj > 0) { struct snd_pcm_runtime *runtime = substream->runtime; - pos_adj = (bdl_pos_adj * runtime->rate + 47999) / 48000; + pos_adj = (pos_adj * runtime->rate + 47999) / 48000; if (!pos_adj) pos_adj = 1; pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { snd_printk(KERN_WARNING "Too big adjustment %d\n", - bdl_pos_adj); + bdl_pos_adj[chip->dev_index]); pos_adj = 0; } else { ofs = setup_bdle(substream, azx_dev, @@ -1063,7 +1066,8 @@ static int azx_setup_periods(struct snd_pcm_substream *substream, goto error; azx_dev->irq_ignore = 1; } - } + } else + pos_adj = 0; for (i = 0; i < periods; i++) { if (i == periods - 1 && pos_adj) ofs = setup_bdle(substream, azx_dev, &bdl, ofs, @@ -1388,7 +1392,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(substream, azx_dev) < 0) + if (azx_setup_periods(chip, substream, azx_dev) < 0) return -EINVAL; azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -2001,6 +2005,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; chip->msi = enable_msi; + chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); chip->position_fix = check_position_fix(chip, position_fix[dev]); -- cgit v1.2.3 From 5c0d7bc103dd1ae85967fbcf70be8d1ae3aa3d79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:35 +0200 Subject: ALSA: hda - Fix bdl_pos_adj value for ATI SB chipsets ATI SB controllers seem to report the DMA ahead in the amount of FIFO. Thus bdl_pos_adj should be 32 for them as default. Also, the default value is set to -1, which means to make the driver to choose the appropriate value. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ddae3c479a88..9b2dc0669b94 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -55,7 +55,7 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static char *model[SNDRV_CARDS]; static int position_fix[SNDRV_CARDS]; -static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1}; +static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int single_cmd; static int enable_msi; @@ -2013,6 +2013,18 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->single_cmd = single_cmd; + if (bdl_pos_adj[dev] < 0) { + switch (chip->driver_type) { + case AZX_DRIVER_ATI: + case AZX_DRIVER_ATIHDMI: + bdl_pos_adj[dev] = 32; + break; + default: + bdl_pos_adj[dev] = 1; + break; + } + } + #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ if (chip->driver_type == AZX_DRIVER_ULI) { -- cgit v1.2.3 From a6a950a8a8e1431e8527809339e089ac926a7d34 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Jun 2008 17:53:35 +0200 Subject: ALSA: hda - Add a warning if pending IRQ is found The pending IRQ handling is a very hackish workaround and should be avoided as much as possible via a larger bdl_pos_adj option value. Put a warning message if this situation occurs so that the user may have a chance to notice that something is wrong. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b2dc0669b94..ec4ae2d94000 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -374,6 +374,7 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + unsigned int irq_pending_warned :1; /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ @@ -1562,6 +1563,14 @@ static void azx_irq_pending_work(struct work_struct *work) struct azx *chip = container_of(work, struct azx, irq_pending_work); int i, pending; + if (!chip->irq_pending_warned) { + printk(KERN_WARNING + "hda-intel: IRQ timing workaround is activated " + "for card #%d. Suggest a bigger bdl_pos_adj.\n", + chip->card->number); + chip->irq_pending_warned = 1; + } + for (;;) { pending = 0; spin_lock_irq(&chip->reg_lock); -- cgit v1.2.3 From 4549915cdf468019d2fd86485dd8d2ab426b8d85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jun 2008 16:27:24 +0200 Subject: ALSA: hda - Fix stac9205_cfg_tbl Sort stac9205_cfg_table in the order of id numbers, and removed the duplicated (obsoleted) entries for 0x01fc and 0x01fd. This doesn't change the driver behavior since the old entries are all secondary. The duplication occured due to commit dfe495d0, and the old entries were introduced by commit ae0a8ed8. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a00bae8..1ef3dd6e8a64 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1754,12 +1754,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f8, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, - "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa, "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, @@ -1770,18 +1766,14 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ff, "Dell Precision M4300", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, - "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fc, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fd, - "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0204, "unknown Dell", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0206, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, + "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, -- cgit v1.2.3 From 149b154452b18dedd4559cdd65b965ae9011e425 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jun 2008 17:43:36 +0200 Subject: ALSA: hda - Remove unused mutex Removed unused mutex from patch_*.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 5 ----- sound/pci/hda/patch_conexant.c | 4 ---- 2 files changed, 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a99e86d74278..55ce897ef0b3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include "hda_codec.h" @@ -64,7 +63,6 @@ struct ad198x_spec { /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -3177,7 +3175,6 @@ static int patch_ad1884(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -3847,7 +3844,6 @@ static int patch_ad1884a(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -4152,7 +4148,6 @@ static int patch_ad1882(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 36fd85260035..e3df8ac37076 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -82,7 +82,6 @@ struct conexant_spec { /* PCM information */ struct hda_pcm pcm_rec[2]; /* used in build_pcms() */ - struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -928,7 +927,6 @@ static int patch_cxt5045(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1477,7 +1475,6 @@ static int patch_cxt5047(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1736,7 +1733,6 @@ static int patch_cxt5051(struct hda_codec *codec) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; codec->patch_ops = conexant_patch_ops; -- cgit v1.2.3 From aafc4412be31306e5c38bd4e4581066507503adc Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Fri, 13 Jun 2008 18:04:33 +0200 Subject: ALSA: hda: Add support for 92HD73xxx codecs Added support for new family of IDT codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 47 +++++++++++++++++++++++++++++++----------- 1 file changed, 35 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ef3dd6e8a64..c4f3489376c5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -636,21 +636,28 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; +#define HD_DISABLE_PORTF 3 static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* start of config #1 */ + + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for node 0x0f */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* start of config #2 */ + /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ + /* connect port 0d to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1317,13 +1324,13 @@ static unsigned int ref92hd71bxx_pin_configs[10] = { 0x90a000f0, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x044413b0, @@ -3095,13 +3102,16 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + /* skip any ports that don't have jacks since presence + * detection is useless */ + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3606,6 +3616,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, @@ -3634,6 +3645,19 @@ again: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; break; + case 0x111d7608: /* 5 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + spec->mixer = stac92hd71bxx_analog_mixer; + + /* disable VSW */ + spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ default: spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; @@ -3652,15 +3676,13 @@ again: spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; @@ -4298,10 +4320,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, -- cgit v1.2.3 From 0c6341ace5efb14a0da08482e299b3c158e0d5fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jun 2008 20:50:27 +0200 Subject: ALSA: hda - bdl_pos_adj=32 as default Use bdl_pos_adj=32 as default except for Intel hardwares confirmed to work with bdl_pos_adj=1. Looks like ATI and NVidia require this higher value. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec4ae2d94000..539d20102c32 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2024,12 +2024,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, if (bdl_pos_adj[dev] < 0) { switch (chip->driver_type) { - case AZX_DRIVER_ATI: - case AZX_DRIVER_ATIHDMI: - bdl_pos_adj[dev] = 32; + case AZX_DRIVER_ICH: + bdl_pos_adj[dev] = 1; break; default: - bdl_pos_adj[dev] = 1; + bdl_pos_adj[dev] = 32; break; } } -- cgit v1.2.3 From 766979e09d302315f314bfd96fdd83f8f9896d9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jun 2008 20:53:56 +0200 Subject: ALSA: hda - use upper_32_bits() Use the standard upper_32_bits() instead of own macro. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_intel.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 539d20102c32..16715a68ba5e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -440,11 +440,6 @@ static char *driver_short_names[] __devinitdata = { /* for pcm support */ #define get_azx_dev(substream) (substream->runtime->private_data) -/* Get the upper 32bit of the given dma_addr_t - * Compiler should optimize and eliminate the code if dma_addr_t is 32bit - */ -#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0) - static int azx_acquire_irq(struct azx *chip, int do_disconnect); /* @@ -475,7 +470,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->corb.addr = chip->rb.addr; chip->corb.buf = (u32 *)chip->rb.area; azx_writel(chip, CORBLBASE, (u32)chip->corb.addr); - azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr)); + azx_writel(chip, CORBUBASE, upper_32_bits(chip->corb.addr)); /* set the corb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, CORBSIZE, 0x02); @@ -490,7 +485,7 @@ static void azx_init_cmd_io(struct azx *chip) chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr); - azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr)); + azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr)); /* set the rirb size to 256 entries (ULI requires explicitly) */ azx_writeb(chip, RIRBSIZE, 0x02); @@ -861,7 +856,7 @@ static void azx_init_chip(struct azx *chip) /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); - azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr)); + azx_writel(chip, DPUBASE, upper_32_bits(chip->posbuf.addr)); chip->initialized = 1; } @@ -1006,7 +1001,7 @@ static int setup_bdle(struct snd_pcm_substream *substream, addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); - bdl[1] = cpu_to_le32(upper_32bit(addr)); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ chunk = PAGE_SIZE - (ofs % PAGE_SIZE); if (size < chunk) @@ -1138,7 +1133,7 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* lower BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ - azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl.addr)); + azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ if (chip->position_fix == POS_FIX_POSBUF || -- cgit v1.2.3 From 6719292af5513da52aa876d0e7ac4367a0851845 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 16 Jun 2008 10:39:34 +0200 Subject: ALSA: Remove duplicate MODULE_AUTHOR/DESCRIPTION/LICENCE from snd-ens1370.ko But comment only extra code in ak4531_codec.c for history. Signed-off-by: Jaroslav Kysela --- sound/pci/ak4531_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 6a99eed2d8b6..33d37b1c42fc 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -28,9 +28,11 @@ #include #include +/* MODULE_AUTHOR("Jaroslav Kysela "); MODULE_DESCRIPTION("Universal routines for AK4531 codec"); MODULE_LICENSE("GPL"); +*/ #ifdef CONFIG_PROC_FS static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak4531); -- cgit v1.2.3 From 9f515b6898d0f2c1b9eb34ae9986794045f34e2b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 17 Jun 2008 16:20:13 +0200 Subject: ALSA: emu10k1 - fix possible memory leak in memory allocation routines The leak was introduced in "[ALSA] emu10k1 - simplify page allocation for synth" commit. Signed-off-by: Jaroslav Kysela --- sound/pci/emu10k1/memory.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 42943b4fcb7b..759e29f89478 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -464,11 +464,17 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk /* first try to allocate from <4GB zone */ struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | __GFP_NOWARN); - if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) + if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | + struct page *p1 = + alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); + /* free page outside dma_mask range */ + if (p) + free_page((unsigned long)page_address(p)); + p = p1; + } if (!p) { __synth_free_pages(emu, first_page, page - 1); return -ENOMEM; -- cgit v1.2.3 From 284373059f9605442cac6453780f6aaecf9abac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jun 2008 16:30:27 +0200 Subject: ALSA: emu10k1 - simplify the last fix Clean up the previous commit for fixing memory leaks. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 759e29f89478..7d379f5131fb 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -465,15 +465,12 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | __GFP_NOWARN); if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { + if (p) + __free_page(p); /* try to allocate from <16MB zone */ - struct page *p1 = - alloc_page(GFP_ATOMIC | GFP_DMA | + p = alloc_page(GFP_ATOMIC | GFP_DMA | __GFP_NORETRY | /* no OOM-killer */ __GFP_NOWARN); - /* free page outside dma_mask range */ - if (p) - free_page((unsigned long)page_address(p)); - p = p1; } if (!p) { __synth_free_pages(emu, first_page, page - 1); -- cgit v1.2.3 From 43785eaeb1cfb8aed3cf8027f298b242f88fdc45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Jun 2008 15:47:26 +0200 Subject: ALSA: hda - Fix wrong volumes in AD1988 auto-probe mode Don't create mixer volume elements for Headphone and Speaker if they use the same DAC as normal line-outs on AD1988. Otherwise the amp value gets screwed up, e.g. https://bugzilla.novell.com/show_bug.cgi?id=398255 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 32 +++++++++++++++++++++----------- 1 file changed, 21 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 55ce897ef0b3..05ca027fcc01 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2621,7 +2621,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, { struct ad198x_spec *spec = codec->spec; hda_nid_t nid; - int idx, err; + int i, idx, err; char name[32]; if (! pin) @@ -2629,16 +2629,26 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx); - /* specify the DAC as the extra output */ - if (! spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else - spec->multiout.extra_out_nid[0] = nid; - /* control HP volume/switch on the output mixer amp */ - sprintf(name, "%s Playback Volume", pfx); - if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) - return err; + /* check whether the corresponding DAC was already taken */ + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t pin = spec->autocfg.line_out_pins[i]; + hda_nid_t dac = ad1988_idx_to_dac(codec, ad1988_pin_idx(pin)); + if (dac == nid) + break; + } + if (i >= spec->autocfg.line_outs) { + /* specify the DAC as the extra output */ + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; + /* control HP volume/switch on the output mixer amp */ + sprintf(name, "%s Playback Volume", pfx); + err = add_control(spec, AD_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } nid = ad1988_mixer_nids[idx]; sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, AD_CTL_BIND_MUTE, name, -- cgit v1.2.3 From f52845ad771a2b62bc06bc940f16c8f6296654ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Jun 2008 16:35:22 +0200 Subject: ALSA: ca0106 - Add entry for another MSI K8N Diamond MB Added an entry for another MSI K8N Diamond mobo with SSID 1102:1009. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ca0106/ca0106_main.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index ecbe79b67e43..2f8b28add276 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -249,6 +249,11 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , + /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + { .serial = 0x10091102, + .name = "MSI K8N Diamond MB", + .gpio_type = 2, + .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- cgit v1.2.3 From a1855d802fb62718192eb7e180161b08adff4e73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jun 2008 15:41:37 +0200 Subject: ALSA: hda - Fix digital converter proc output AC_VERB_GET_DIGI_CONVERT_2 isn't actually implemented but reserved. The whole SIC bits are returned from DIGI_CONVERT_1. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_proc.c | 5 ++--- 2 files changed, 3 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index dcd390b2bbaa..efc682888b31 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -78,7 +78,7 @@ enum { #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d -#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 5633f77f8f3b..1e5aff5c48d1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -366,8 +366,6 @@ static void print_digital_conv(struct snd_info_buffer *buffer, { unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); - unsigned int digi2 = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_2, 0); snd_iprintf(buffer, " Digital:"); if (digi1 & AC_DIG1_ENABLE) snd_iprintf(buffer, " Enabled"); @@ -386,7 +384,8 @@ static void print_digital_conv(struct snd_info_buffer *buffer, if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); snd_iprintf(buffer, "\n"); - snd_iprintf(buffer, " Digital category: 0x%x\n", digi2 & AC_DIG2_CC); + snd_iprintf(buffer, " Digital category: 0x%x\n", + (digi1 >> 8) & AC_DIG2_CC); } static const char *get_pwr_state(u32 state) -- cgit v1.2.3 From 3e0e469fa216ec70c93b1593821b759d19ee2e6b Mon Sep 17 00:00:00 2001 From: Travis Place Date: Fri, 20 Jun 2008 16:51:45 +0200 Subject: ALSA: hda - Added model selection for iMac 24" Added the SSID of a known iMac 24" to automatically use ALC885_IMAC24 quirk. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 61f8c13b2bb9..d96a8762b287 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6192,6 +6192,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), -- cgit v1.2.3 From 1d11604eebfc91204e3182b48d3d920f51b09d40 Mon Sep 17 00:00:00 2001 From: Travis Place Date: Mon, 23 Jun 2008 11:42:30 +0200 Subject: ALSA: hda - Added SSID for 'Fujitsu Siemens Amilo M1451G' laptop Add the SSID for the "Fujitsu Siemens Amilo M1451G" laptop to patch_realtek.c , so that it uses ALC880_FUJITSU by default. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d96a8762b287..4edde6094cd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3074,6 +3074,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), -- cgit v1.2.3 From 981bcead3f2279a1ec6fb5f2c57aff79ed61a700 Mon Sep 17 00:00:00 2001 From: Pierre Ossman Date: Mon, 23 Jun 2008 11:45:47 +0200 Subject: ALSA: trident - pause s/pdif output Stop the S/PDIF DMA engine and output when the device is told to pause. It will keep on looping the current buffer contents if this isn't done. Signed-off-by: Pierre Ossman Tested-by: Rene Herman Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/trident/trident_main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index bbcee2c09ae4..a69b4206c69e 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -1590,7 +1590,10 @@ static int snd_trident_trigger(struct snd_pcm_substream *substream, if (spdif_flag) { if (trident->device != TRIDENT_DEVICE_ID_SI7018) { outl(trident->spdif_pcm_bits, TRID_REG(trident, NX_SPCSTATUS)); - outb(trident->spdif_pcm_ctrl, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); + val = trident->spdif_pcm_ctrl; + if (!go) + val &= ~(0x28); + outb(val, TRID_REG(trident, NX_SPCTRL_SPCSO + 3)); } else { outl(trident->spdif_pcm_bits, TRID_REG(trident, SI_SPDIF_CS)); val = inl(TRID_REG(trident, SI_SERIAL_INTF_CTRL)) | SPDIF_EN; -- cgit v1.2.3 From 627d3e7abca30d6e86787c98dd7cbac0233bc5a9 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Mon, 23 Jun 2008 11:50:47 +0200 Subject: ALSA: PCI168 snd-azt3328: some more fixups - fix problem with codec register 0x6a being write-only by adding a software shadow register (caused annoying noise after module loading due to _toggling_ between gameport and audio bits instead of configuring them properly) - rename several "Wave" mixer controls to "PCM", since this is what Wine and several other apps are looking for (IOW, _requiring_) and this is what AC97 specs use as naming, too, thus I'd guess it's what these controls are - cleanup, small optimizations Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/azt3328.c | 109 ++++++++++++++++++++++++++++++---------------------- sound/pci/azt3328.h | 16 ++++++-- 2 files changed, 76 insertions(+), 49 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index b832333c3023..22f18f3cfbc9 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -289,6 +289,12 @@ struct snd_azf3328 { struct pci_dev *pci; int irq; + /* register 0x6a is write-only, thus need to remember setting. + * If we need to add more registers here, then we might try to fold this + * into some transparent combined shadow register handling with + * CONFIG_PM register storage below, but that's slightly difficult. */ + u16 shadow_reg_codec_6AH; + #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ @@ -324,21 +330,6 @@ snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) return 0; } -static int -snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) -{ - u16 prev = inw(reg), new; - - new = (do_set) ? (prev|mask) : (prev & ~mask); - /* we need to always write the new value no matter whether it differs - * or not, since some register bits don't indicate their setting */ - outw(new, reg); - if (new != prev) - return 1; - - return 0; -} - static inline void snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { @@ -662,7 +653,7 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; - const char *p = NULL; + const char * const *p = NULL; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -673,20 +664,20 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol, if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - p = texts1[uinfo->value.enumerated.item]; + p = texts1; break; case 9: /* mono sel source */ - p = texts2[uinfo->value.enumerated.item]; + p = texts2; break; case 15: /* PCM Out Path */ - p = texts4[uinfo->value.enumerated.item]; + p = texts4; break; } } else if (reg.reg == IDX_MIXER_REC_SELECT) - p = texts3[uinfo->value.enumerated.item]; + p = texts3; - strcpy(uinfo->value.enumerated.name, p); + strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); return 0; } @@ -745,9 +736,11 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1), AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), - AZF3328_MIXER_VOL_STEREO("Wave Playback Volume", IDX_MIXER_WAVEOUT, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave 3D Bypass Playback Switch", IDX_MIXER_ADVCTL2, 7, 1), + AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), + AZF3328_MIXER_VOL_STEREO("PCM Playback Volume", + IDX_MIXER_WAVEOUT, 0x1f, 1), + AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch", + IDX_MIXER_ADVCTL2, 7, 1), AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1), AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1), AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1), @@ -874,7 +867,7 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, unsigned reg, - unsigned int bitrate, + enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) @@ -958,15 +951,29 @@ snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); } +static void +snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, + unsigned bitmask, + int enable +) +{ + if (enable) + chip->shadow_reg_codec_6AH &= ~bitmask; + else + chip->shadow_reg_codec_6AH |= bitmask; + snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", + bitmask, enable, chip->shadow_reg_codec_6AH); + snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); +} + static inline void snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) { + snd_azf3328_dbgplay("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_PAUSE_PLAYBACK_BIT8, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } @@ -1404,10 +1411,8 @@ snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) static inline void snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) { - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_SOMETHING2_GAMEPORT, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1525,8 +1530,6 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) { struct gameport *gp; - int io_port = chip->game_io; - chip->gameport = gp = gameport_allocate_port(); if (!gp) { printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); @@ -1536,7 +1539,7 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) gameport_set_name(gp, "AZF3328 Gameport"); gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = io_port; + gp->io = chip->game_io; gameport_set_port_data(gp, chip); gp->open = snd_azf3328_gameport_open; @@ -1577,6 +1580,15 @@ snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) /******************************************************************/ +static inline void +snd_azf3328_irq_log_unknown_type(u8 which) +{ + snd_azf3328_dbgplay( + "azt3328: unknown IRQ type (%x) occurred, please report!\n", + which + ); +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { @@ -1594,11 +1606,14 @@ snd_azf3328_interrupt(int irq, void *dev_id) )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), - status); + snd_azf3328_dbgplay( + "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " + "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + irq_count++ /* debug-only */, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), + status + ); if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", @@ -1631,9 +1646,9 @@ snd_azf3328_interrupt(int irq, void *dev_id) ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); @@ -1653,9 +1668,9 @@ snd_azf3328_interrupt(int irq, void *dev_id) ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -2311,6 +2326,10 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + + /* manually store the one currently relevant write-only reg, too */ + chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 3448fd626f80..7e3e8942d073 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -1,7 +1,8 @@ #ifndef __SOUND_AZT3328_H #define __SOUND_AZT3328_H -/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */ +/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 + * "WRITE_ONLY" == register does not indicate actual bit values */ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ @@ -76,7 +77,7 @@ #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 /* define frequency helpers, for maximum value safety */ -enum { +enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate AZF_FREQ(4000), AZF_FREQ(4800), @@ -150,11 +151,18 @@ enum { #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ /* umm, nope, behaviour of these bits changes depending on what we wrote - * to 0x6b!! */ + * to 0x6b!! + * And they change upon playback/stop, too: + * Writing a value to 0x68 will display this exact value during playback, + * too but when stopped it can fall back to a rather different + * seemingly random value). Hmm, possibly this is a register which + * has a remote shadow which needs proper device supply which only exists + * in case playback is active? Or is this driver-induced? + */ /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); * actually inhibits PCM playback!!! maybe power management??: */ -#define IDX_IO_6AH 0x6A +#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */ /* bit 5: enabling this will activate permanent counting of bytes 2/3 * at gameport I/O (0xb402/3) (equal values each) and cause * gameport legacy I/O at 0x0200 to be _DISABLED_! -- cgit v1.2.3 From c7e0757a461db87157bbf92c4f5afda87cbe05f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jun 2008 14:42:51 +0200 Subject: ALSA: hda - Add MacBook 3.1 support MacBook 3.1 is compatible with model=mbp3. Added the codec SSID 10b6:3600. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4edde6094cd1..c7d4bbb01798 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6519,8 +6519,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b1000: /* iMac 24 */ board_config = ALC885_IMAC24; break; - case 0x106b00a1: /* Macbook */ + case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b2c00: /* Macbook Pro rev3 */ + case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: -- cgit v1.2.3 From 4a3b6983232cd296ea260e06461d031f10065d63 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jun 2008 17:17:00 +0200 Subject: ALSA: ymfpci - fix initial volume for 44.1kHz output YDSXGR_BUF441OUTVOL register isn't initialized properly. This may lead to a silent output at full volume of unchanged "Wave Playback Volume". http://bugzilla.kernel.org/show_bug.cgi?id=10963 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ymfpci/ymfpci_main.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 29b3056c5109..7129df5f315b 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2205,6 +2205,7 @@ static int __devinit snd_ymfpci_memalloc(struct snd_ymfpci *chip) for (reg = 0x80; reg < 0xc0; reg += 4) snd_ymfpci_writel(chip, reg, 0); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x3fff3fff); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_ZVOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_SPDIFOUTVOL, 0x3fff3fff); snd_ymfpci_writel(chip, YDSXGR_NATIVEADCINVOL, 0x3fff3fff); @@ -2324,6 +2325,7 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); + snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); pci_disable_device(pci); pci_save_state(pci); -- cgit v1.2.3 From 031005f78c8d0aebc17ddf7a34af9ffd48034d7d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jun 2008 14:49:58 +0200 Subject: ALSA: hda - disable amp override on non-HP machines Some machines with Cx5045 seem to have no problem with the volume over 0dB on NID 0x17. Disable amp overrides for other devices but HP. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e3df8ac37076..64df8100593d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1005,15 +1005,19 @@ static int patch_cxt5045(struct hda_codec *codec) #endif } - /* - * Fix max PCM level to 0 dB - * (originall it has 0x2b steps with 0dB offset 0x14) - */ - snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, - (0x14 << AC_AMPCAP_OFFSET_SHIFT) | - (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptop has a really bad sound over 0dB on NID 0x17. + * Fix max PCM level to 0 dB + * (originall it has 0x2b steps with 0dB offset 0x14) + */ + snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, + (0x14 << AC_AMPCAP_OFFSET_SHIFT) | + (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; + } return 0; } -- cgit v1.2.3 From 942fd1ebf6072fdaa07ae6d77a4f58b39c1bfdf9 Mon Sep 17 00:00:00 2001 From: Walter Sheets Date: Fri, 27 Jun 2008 11:53:31 +0200 Subject: ALSA: via82xx - Add VIA audio device #1841 to ac97_quirk list Signed-off-by: Walter Sheets Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/via82xx.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3e4c47..6781be9e3078 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1756,6 +1756,12 @@ static struct ac97_quirk ac97_quirks[] = { .name = "ECS L7VMM2", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x1019, + .subdevice = 0x1841, + .name = "ECS K7VTA3", + .type = AC97_TUNE_HP_ONLY + }, { .subvendor = 0x1849, .subdevice = 0x3059, -- cgit v1.2.3 From 6cabf6b0694951288520512e0b68c78523acc9fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jun 2008 11:55:20 +0200 Subject: ALSA: Fix AC97 power down Some laptops don't like PR3 powerdown. Do PR3 powerdown only for the real power-saving. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_codec.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 2d2f16e11082..07364c00768a 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2295,9 +2295,11 @@ static void snd_ac97_powerdown(struct snd_ac97 *ac97) power |= AC97_PD_PR0 | AC97_PD_PR1; /* ADC & DAC powerdown */ snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); - power |= AC97_PD_PR2 | AC97_PD_PR3; /* Analog Mixer powerdown */ + power |= AC97_PD_PR2; /* Analog Mixer powerdown (Vref on) */ snd_ac97_write(ac97, AC97_POWERDOWN, power); if (ac97_is_power_save_mode(ac97)) { + power |= AC97_PD_PR3; /* Analog Mixer powerdown */ + snd_ac97_write(ac97, AC97_POWERDOWN, power); udelay(100); /* AC-link powerdown, internal Clk disable */ /* FIXME: this may cause click noises on some boards */ -- cgit v1.2.3 From 0701e0640f4a1de2b509cfee508216275f81d812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jun 2008 16:30:57 +0200 Subject: ALSA: ALSA: hda - Fix ALC883 medion model ALC883 medion model doesn't unmute the proper amps so no output can be heard. Replaced the mute switches to behave just like other models. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7d4bbb01798..081d1a557065 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7050,13 +7050,13 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), -- cgit v1.2.3 From 470eaf6be78424fc499a5039e5d5fe58bace2bc3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jun 2008 16:40:10 +0200 Subject: ALSA: hda - Add missing Thinkpad Z60m support Added the missing SSID of Thinkpad Z60m for model=thinkpad with AD1981HD. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 05ca027fcc01..e8003d99f0bf 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1616,6 +1616,7 @@ static const char *ad1981_models[AD1981_MODELS] = { static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), + SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), -- cgit v1.2.3 From 6a9dccd61253f361760802d8d19c28fa83ea83f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 1 Jul 2008 14:52:05 +0200 Subject: ALSA: ALSA: hda - Fix speaker output on Toshiba P105 Toshiba Satellite P105 with cx5045 has no HP pin but only a speaker pin and does the speaker-muting on hardware. Thus the matching model is laptop-micsense. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 64df8100593d..95e3367d8879 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -906,6 +906,7 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), -- cgit v1.2.3 From 25552e87aa4102f048f1a8a8ddc87c96593c304e Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Wed, 2 Jul 2008 13:19:23 +0300 Subject: ALSA: ac97 - fix patch_ucb1400 for proper resume Replace 'snd_ac97_write' with snd_ac97_write_cache' in pacth_ucb1400 to allow proper codec wakeup. Signed-off-by: Mike Rapoport Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index abe88adec37e..0746e9ccc20b 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3710,7 +3710,7 @@ static int patch_ucb1400(struct snd_ac97 * ac97) { ac97->build_ops = &patch_ucb1400_ops; /* enable headphone driver and smart low power mode by default */ - snd_ac97_write(ac97, 0x6a, 0x0050); - snd_ac97_write(ac97, 0x6c, 0x0030); + snd_ac97_write_cache(ac97, 0x6a, 0x0050); + snd_ac97_write_cache(ac97, 0x6c, 0x0030); return 0; } -- cgit v1.2.3 From be321a890c25272965129ffe4b3b59a519fcf583 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 Jul 2008 16:04:04 +0200 Subject: ALSA: hda - Add support for Lenovo 3000 N200 Added the model entry (model=lenovo) for Lenovo N3000 N200 laptop with ALC861-VD. Reference below: https://bugzilla.novell.com/show_bug.cgi?id=406425 Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 081d1a557065..92a709be3c4e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13004,6 +13004,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; -- cgit v1.2.3 From 7e2574050e6af203f0c94915c98c53ce1fc48044 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Tue, 8 Jul 2008 11:50:09 +0200 Subject: ALSA: hda - removed redundant gpio_mask An gpio_mask value was defined twice needlessly. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c4f3489376c5..a6d138831e26 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3669,7 +3669,6 @@ again: /* GPIO0 High = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; - spec->gpio_mask = 0x01; spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids; -- cgit v1.2.3 From 34ec8a0ae138c2787a550b930d29a9cce4900cee Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 10 Jul 2008 14:49:19 +0200 Subject: ALSA: HDA - HP dc7600 with pci sub IDs 0x103c/0x3011 belongs to hp-3013 model As reported and tested by an RedHat customer, HP dc7600 with pci sub IDs 0x103c/0x3011 works with the hp-3013 model and not with the hp only model. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 92a709be3c4e..2807bc840d26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5167,7 +5167,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP), + SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP), -- cgit v1.2.3 From c2ba47d776bf9a45e15f28fc73ad44877437bef9 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Sat, 24 May 2008 00:01:40 +0100 Subject: vx222: treat firmware data as const Signed-off-by: David Woodhouse --- sound/pci/vx222/vx222_ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index b4bfc1acde88..631f3a639993 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -359,7 +359,7 @@ static int vx2_load_xilinx_binary(struct vx_core *chip, const struct firmware *x { unsigned int i; unsigned int port; - unsigned char *image; + const unsigned char *image; /* XILINX reset (wait at least 1 milisecond between reset on and off). */ vx_outl(chip, CNTRL, VX_CNTRL_REGISTER_VALUE | VX_XILINX_RESET_MASK); -- cgit v1.2.3 From 93a9c901c88ba2b1bae9dd55e6243896b8a580f1 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Sat, 24 May 2008 00:02:03 +0100 Subject: riptide: treat firmware data as const Signed-off-by: David Woodhouse --- sound/pci/riptide/riptide.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 979f7da641ce..6a3596247348 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -682,7 +682,7 @@ static union firmware_version firmware_versions[] = { }, }; -static u32 atoh(unsigned char *in, unsigned int len) +static u32 atoh(const unsigned char *in, unsigned int len) { u32 sum = 0; unsigned int mult = 1; @@ -702,12 +702,12 @@ static u32 atoh(unsigned char *in, unsigned int len) return sum; } -static int senddata(struct cmdif *cif, unsigned char *in, u32 offset) +static int senddata(struct cmdif *cif, const unsigned char *in, u32 offset) { u32 addr; u32 data; u32 i; - unsigned char *p; + const unsigned char *p; i = atoh(&in[1], 2); addr = offset + atoh(&in[3], 4); @@ -726,10 +726,10 @@ static int senddata(struct cmdif *cif, unsigned char *in, u32 offset) return 0; } -static int loadfirmware(struct cmdif *cif, unsigned char *img, +static int loadfirmware(struct cmdif *cif, const unsigned char *img, unsigned int size) { - unsigned char *in; + const unsigned char *in; u32 laddr, saddr, t, val; int err = 0; -- cgit v1.2.3 From b8d21807a1a479e0214a03069a88e3e93492b72d Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Sat, 24 May 2008 00:02:28 +0100 Subject: pcxhr: treat firmware data as const Signed-off-by: David Woodhouse --- sound/pci/pcxhr/pcxhr_core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 78aa81feaa4a..957e6afe94f1 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -269,7 +269,7 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin unsigned int chipsc; unsigned char data; unsigned char mask; - unsigned char *image; + const unsigned char *image; /* test first xilinx */ chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC); @@ -316,7 +316,7 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp) int err; unsigned int i; unsigned int len; - unsigned char *data; + const unsigned char *data; unsigned char dummy; /* check the length of boot image */ snd_assert(dsp->size > 0, return -EINVAL); -- cgit v1.2.3 From fa6e1cb66e2f9d2d4703e7bd7dd50839bb10e4c3 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Thu, 29 May 2008 11:58:27 +0300 Subject: maestro3: treat firmware data as const The maestro3 driver is byte-swapping its firmware to be host-endian in advance, when it doesn't seem to be necessary -- we could just use le16_to_cpu() as we load it. Doing that means that we need to switch the in-tree firmware to be little-endian too. Take the least intrusive way of doing this, which is to switch the existing snd_m3_convert_from_le() function to convert _to_ little-endian instead, and use it on the in-tree firmware instead of the loaded firmware. It's a bit suboptimal but doesn't matter much right now because we're about to remove the special cases for the in-tree version anyway. Signed-off-by: David Woodhouse --- sound/pci/maestro3.c | 29 +++++++++++++++-------------- 1 file changed, 15 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index a536c59fbea1..9dfba6eff858 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2240,18 +2240,16 @@ static const struct firmware assp_minisrc = { .size = sizeof assp_minisrc_image }; -#else /* CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL */ - #ifdef __LITTLE_ENDIAN -static inline void snd_m3_convert_from_le(const struct firmware *fw) { } +static inline void snd_m3_convert_to_le(const struct firmware *fw) { } #else -static void snd_m3_convert_from_le(const struct firmware *fw) +static void snd_m3_convert_to_le(const struct firmware *fw) { int i; u16 *data = (u16 *)fw->data; for (i = 0; i < fw->size / 2; ++i) - le16_to_cpus(&data[i]); + cpu_to_le16s(&data[i]); } #endif @@ -2271,7 +2269,7 @@ static const u16 minisrc_lpf[MINISRC_LPF_LEN] = { static void snd_m3_assp_init(struct snd_m3 *chip) { unsigned int i; - u16 *data; + const u16 *data; /* zero kernel data */ for (i = 0; i < (REV_B_DATA_MEMORY_UNIT_LENGTH * NUM_UNITS_KERNEL_DATA) / 2; i++) @@ -2289,10 +2287,11 @@ static void snd_m3_assp_init(struct snd_m3 *chip) KDATA_DMA_XFER0); /* write kernel into code memory.. */ - data = (u16 *)chip->assp_kernel_image->data; + data = (const u16 *)chip->assp_kernel_image->data; for (i = 0 ; i * 2 < chip->assp_kernel_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, - REV_B_CODE_MEMORY_BEGIN + i, data[i]); + REV_B_CODE_MEMORY_BEGIN + i, + le16_to_cpu(data[i])); } /* @@ -2301,10 +2300,10 @@ static void snd_m3_assp_init(struct snd_m3 *chip) * drop it there. It seems that the minisrc doesn't * need vectors, so we won't bother with them.. */ - data = (u16 *)chip->assp_minisrc_image->data; + data = (const u16 *)chip->assp_minisrc_image->data; for (i = 0; i * 2 < chip->assp_minisrc_image->size; i++) { snd_m3_assp_write(chip, MEMTYPE_INTERNAL_CODE, - 0x400 + i, data[i]); + 0x400 + i, le16_to_cpu(data[i])); } /* @@ -2749,8 +2748,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) { snd_m3_free(chip); return err; - } else - snd_m3_convert_from_le(chip->assp_kernel_image); + } #endif #ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL @@ -2761,8 +2759,7 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, if (err < 0) { snd_m3_free(chip); return err; - } else - snd_m3_convert_from_le(chip->assp_minisrc_image); + } #endif if ((err = pci_request_regions(pci, card->driver)) < 0) { @@ -2915,6 +2912,10 @@ static struct pci_driver driver = { static int __init alsa_card_m3_init(void) { +#ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL + snd_m3_convert_to_le(&assp_kernel); + snd_m3_convert_to_le(&assp_minisrc); +#endif return pci_register_driver(&driver); } -- cgit v1.2.3 From b82a82d0a90af74847ae3e873a241dedf3786fd5 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Thu, 29 May 2008 14:40:00 +0300 Subject: ymfpci: treat firmware data as const Standardise both in-kernel and loaded firmware to be stored as little-endian instead of host-endian. Signed-off-by: David Woodhouse --- sound/pci/ymfpci/ymfpci_main.c | 59 ++++++++++++++++++++++++------------------ 1 file changed, 34 insertions(+), 25 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 29b3056c5109..6298b29c66bb 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include @@ -2009,11 +2010,34 @@ static struct firmware snd_ymfpci_controller_1e_microcode = { .size = YDSXG_CTRLLENGTH, .data = (u8 *)CntrlInst1E, }; + +#ifdef __BIG_ENDIAN +static int microcode_swapped; +static DEFINE_MUTEX(microcode_swap); + +static void snd_ymfpci_convert_to_le(const struct firmware *fw) +{ + int i; + u32 *data = (u32 *)fw->data; + + for (i = 0; i < fw->size / 4; ++i) + cpu_to_le32s(&data[i]); +} #endif -#ifdef CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) { +#ifdef __BIG_ENDIAN + mutex_lock(µcode_swap); + if (!microcode_swapped) { + snd_ymfpci_convert_to_le(&snd_ymfpci_dsp_microcode); + snd_ymfpci_convert_to_le(&snd_ymfpci_controller_1e_microcode); + snd_ymfpci_convert_to_le(&snd_ymfpci_controller_microcode); + microcode_swapped = 1; + } + mutex_unlock(µcode_swap); +#endif + chip->dsp_microcode = &snd_ymfpci_dsp_microcode; if (chip->device_id == PCI_DEVICE_ID_YAMAHA_724F || chip->device_id == PCI_DEVICE_ID_YAMAHA_740C || @@ -2029,19 +2053,6 @@ static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) #else /* use fw_loader */ -#ifdef __LITTLE_ENDIAN -static inline void snd_ymfpci_convert_from_le(const struct firmware *fw) { } -#else -static void snd_ymfpci_convert_from_le(const struct firmware *fw) -{ - int i; - u32 *data = (u32 *)fw->data; - - for (i = 0; i < fw->size / 4; ++i) - le32_to_cpus(&data[i]); -} -#endif - static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) { int err, is_1e; @@ -2050,9 +2061,7 @@ static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) err = request_firmware(&chip->dsp_microcode, "yamaha/ds1_dsp.fw", &chip->pci->dev); if (err >= 0) { - if (chip->dsp_microcode->size == YDSXG_DSPLENGTH) - snd_ymfpci_convert_from_le(chip->dsp_microcode); - else { + if (chip->dsp_microcode->size != YDSXG_DSPLENGTH) { snd_printk(KERN_ERR "DSP microcode has wrong size\n"); err = -EINVAL; } @@ -2067,9 +2076,7 @@ static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) err = request_firmware(&chip->controller_microcode, name, &chip->pci->dev); if (err >= 0) { - if (chip->controller_microcode->size == YDSXG_CTRLLENGTH) - snd_ymfpci_convert_from_le(chip->controller_microcode); - else { + if (chip->controller_microcode->size != YDSXG_CTRLLENGTH) { snd_printk(KERN_ERR "controller microcode" " has wrong size\n"); err = -EINVAL; @@ -2090,7 +2097,7 @@ static void snd_ymfpci_download_image(struct snd_ymfpci *chip) { int i; u16 ctrl; - u32 *inst; + const __le32 *inst; snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0x00000000); snd_ymfpci_disable_dsp(chip); @@ -2105,14 +2112,16 @@ static void snd_ymfpci_download_image(struct snd_ymfpci *chip) snd_ymfpci_writew(chip, YDSXGR_GLOBALCTRL, ctrl & ~0x0007); /* setup DSP instruction code */ - inst = (u32 *)chip->dsp_microcode->data; + inst = (const __le32 *)chip->dsp_microcode->data; for (i = 0; i < YDSXG_DSPLENGTH / 4; i++) - snd_ymfpci_writel(chip, YDSXGR_DSPINSTRAM + (i << 2), inst[i]); + snd_ymfpci_writel(chip, YDSXGR_DSPINSTRAM + (i << 2), + le32_to_cpu(inst[i])); /* setup control instruction code */ - inst = (u32 *)chip->controller_microcode->data; + inst = (const __le32 *)chip->controller_microcode->data; for (i = 0; i < YDSXG_CTRLLENGTH / 4; i++) - snd_ymfpci_writel(chip, YDSXGR_CTRLINSTRAM + (i << 2), inst[i]); + snd_ymfpci_writel(chip, YDSXGR_CTRLINSTRAM + (i << 2), + le32_to_cpu(inst[i])); snd_ymfpci_enable_dsp(chip); } -- cgit v1.2.3 From 67852dc08c0782735d48ce1e2a6eb44cd02a6ff7 Mon Sep 17 00:00:00 2001 From: David Howells Date: Tue, 8 Jul 2008 17:45:58 +0100 Subject: Fix a const pointer usage warning in the Digigram pcxhr soundcard driver Fix a const pointer usage warning in the Digigram pcxhr compatible soundcard driver. A const pointer is being passed to copy_from_user() to load the firmware into. This is okay in this case because the function has allocated the firmware struct itself, but the const qualifier is part of the firmware struct - so the patch casts the const away. Signed-off-by: David Howells Signed-off-by: David Woodhouse --- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index e6a4bfbb91bb..d2f043278cf4 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -394,7 +394,7 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, (unsigned long)fw.size); return -ENOMEM; } - if (copy_from_user(fw.data, dsp->image, dsp->length)) { + if (copy_from_user((void *)fw.data, dsp->image, dsp->length)) { vfree(fw.data); return -EFAULT; } -- cgit v1.2.3 From fd4f80de4612cc5255c108a8c13df88f89c46654 Mon Sep 17 00:00:00 2001 From: David Howells Date: Tue, 8 Jul 2008 17:43:01 +0100 Subject: Fix a const pointer usage warning in the Digigram miXart soundcard driver Fix a const pointer usage warning in the Digigram miXart soundcard driver. A const pointer is being passed to copy_from_user() to load the firmware into. This is okay in this case because the function has allocated the firmware struct itself, but the const qualifier is part of the firmware struct - so the patch casts the const away. Signed-off-by: David Howells Signed-off-by: David Woodhouse --- sound/pci/mixart/mixart_hwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 122c28efc483..f98603146132 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -613,7 +613,7 @@ static int mixart_hwdep_dsp_load(struct snd_hwdep *hw, (int)dsp->length); return -ENOMEM; } - if (copy_from_user(fw.data, dsp->image, dsp->length)) { + if (copy_from_user((void *) fw.data, dsp->image, dsp->length)) { vfree(fw.data); return -EFAULT; } -- cgit v1.2.3 From 76770664dcbc008300c2ac8747671efcc4f78c2d Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Mon, 26 May 2008 23:01:27 +0100 Subject: firmware: convert korg1212 driver to use firmware loader exclusively Signed-off-by: David Woodhouse --- sound/pci/Kconfig | 10 - sound/pci/korg1212/korg1212-firmware.h | 987 --------------------------------- sound/pci/korg1212/korg1212.c | 18 - 3 files changed, 1015 deletions(-) delete mode 100644 sound/pci/korg1212/korg1212-firmware.h (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7e4742109572..1abbf877f20d 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -734,7 +734,6 @@ config SND_INTEL8X0M config SND_KORG1212 tristate "Korg 1212 IO" depends on SND - select FW_LOADER if !SND_KORG1212_FIRMWARE_IN_KERNEL select SND_PCM help Say Y here to include support for Korg 1212IO soundcards. @@ -742,15 +741,6 @@ config SND_KORG1212 To compile this driver as a module, choose M here: the module will be called snd-korg1212. -config SND_KORG1212_FIRMWARE_IN_KERNEL - bool "In-kernel firmware for Korg1212 driver" - depends on SND_KORG1212 - default y - help - Say Y here to include the static firmware built in the kernel - for the Korg1212 driver. If you choose N here, you need to - install the firmware files from the alsa-firmware package. - config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" depends on SND diff --git a/sound/pci/korg1212/korg1212-firmware.h b/sound/pci/korg1212/korg1212-firmware.h deleted file mode 100644 index f6f5b91806a8..000000000000 --- a/sound/pci/korg1212/korg1212-firmware.h +++ /dev/null @@ -1,987 +0,0 @@ -static char dspCode [] = { -0x01,0xff,0x18,0xff,0xf5,0xff,0xcf,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x26,0xff,0x18,0xff,0xff,0xff,0x0f,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x00,0xff,0x0a,0xff,0xff,0xff,0x1f,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, -0x00,0xff,0x0a,0xff,0xff,0xff,0x1f,0xff,0x00,0xff,0x00,0xff,0xff,0xff,0x00,0xff, 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-0x00,0xff,0x68,0xff,0xff,0xff,0x05,0xff,0x00,0xff,0x68,0xff,0xff,0xff,0x45,0xff, -0x00,0xff,0x0a,0xff,0xff,0xff,0x0f,0xff,0x86,0xff,0x87,0xff,0xff,0xff,0xf0,0xff, -0x86,0xff,0x8b,0xff,0xff,0xff,0xe0,0xff,0x00,0xff,0x34,0xff,0xff,0xff,0xc8,0xff, -0x00,0xff,0x38,0xff,0xff,0xff,0xc8,0xff,0x00,0xff,0x60,0xff,0xff,0xff,0x03,0xff, -0x00,0xff,0x60,0xff,0xff,0xff,0x13,0xff,0x00,0xff,0x78,0xff,0xff,0xff,0x13,0xff, -0x00,0xff,0x78,0xff,0xff,0xff,0x03,0xff,0x86,0xff,0x97,0xff,0xff,0xff,0xf0,0xff, -0x86,0xff,0x9b,0xff,0xff,0xff,0xe0,0xff,0x05,0xff,0x81,0xff,0xff,0xff,0xc0,0xff, -0x78,0xff,0x22,0xff,0xff,0xff,0x0f,0xff,0x21,0xff,0x18,0xff,0xff,0xff,0x71,0xff, -0x00,0xff,0x0a,0xff,0xff,0xff,0x0f,0xff,0x7f,0xff,0x38,0xff,0xff,0xff,0x01,0xff, -0x00,0xff,0x38,0xff,0xff,0xff,0x09,0xff,0x00,0xff,0x38,0xff,0xff,0xff,0x06,0xff, -0x7e,0xff,0x40,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x40,0xff,0xff,0xff,0xb1,0xff, -0x08,0xff,0x0c,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x3c,0xff,0xff,0xff,0xc5,0xff, -0x7a,0xff,0x14,0xff,0xff,0xff,0xbe,0xff,0x00,0xff,0x50,0xff,0xff,0xff,0x46,0xff, -0xe1,0xff,0x22,0xff,0xff,0xff,0x0f,0xff,0x7a,0xff,0x1c,0xff,0xff,0xff,0xe0,0xff, -0x60,0xff,0x22,0xff,0xff,0xff,0x0f,0xff,0x00,0xff,0x58,0xff,0xff,0xff,0xa7,0xff, -0x0c,0xff,0x0c,0xff,0xff,0xff,0x00,0xff,0x00,0xff,0x0a,0xff,0xff,0xff,0x0f,0xff, -0x00,0xff,0x40,0xff,0xff,0xff,0xc4,0xff,0x00,0xff,0x0a,0xff,0xff,0xff,0x0f,0xff, -0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff,0xff, -0xff,0xff,0xff,0xff }; diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index f4c85b52bde3..4a44c0f20f76 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -260,14 +260,6 @@ enum MonitorModeSelector { #define COMMAND_ACK_DELAY 13 // number of RTC ticks to wait for an acknowledgement // from the card after sending a command. -#ifdef CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL -#include "korg1212-firmware.h" -static const struct firmware static_dsp_code = { - .data = (u8 *)dspCode, - .size = sizeof dspCode -}; -#endif - enum ClockSourceIndex { K1212_CLKIDX_AdatAt44_1K = 0, // selects source as ADAT at 44.1 kHz K1212_CLKIDX_AdatAt48K, // selects source as ADAT at 48 kHz @@ -412,9 +404,7 @@ struct snd_korg1212 { MODULE_DESCRIPTION("korg1212"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{KORG,korg1212}}"); -#ifndef CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL MODULE_FIRMWARE("korg/k1212.dsp"); -#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -2348,9 +2338,6 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * korg1212->AdatTimeCodePhy = korg1212->sharedBufferPhy + offsetof(struct KorgSharedBuffer, AdatTimeCode); -#ifdef CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL - dsp_code = &static_dsp_code; -#else err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { release_firmware(dsp_code); @@ -2358,15 +2345,12 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * snd_korg1212_free(korg1212); return err; } -#endif if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), dsp_code->size, &korg1212->dma_dsp) < 0) { snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); snd_korg1212_free(korg1212); -#ifndef CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL release_firmware(dsp_code); -#endif return -ENOMEM; } @@ -2376,9 +2360,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size); -#ifndef CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL release_firmware(dsp_code); -#endif rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); -- cgit v1.2.3 From a292f404fabb342716a9d96e8155b7fb7b651dc1 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Thu, 29 May 2008 14:48:34 +0300 Subject: firmware: convert maestro3 driver to use firmware loader exclusively Signed-off-by: David Woodhouse --- sound/pci/Kconfig | 10 --- sound/pci/maestro3.c | 171 --------------------------------------------------- 2 files changed, 181 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 1abbf877f20d..32836ea45170 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -744,7 +744,6 @@ config SND_KORG1212 config SND_MAESTRO3 tristate "ESS Allegro/Maestro3" depends on SND - select FW_LOADER if !SND_MAESTRO3_FIRMWARE_IN_KERNEL select SND_AC97_CODEC help Say Y here to include support for soundcards based on ESS Maestro 3 @@ -753,15 +752,6 @@ config SND_MAESTRO3 To compile this driver as a module, choose M here: the module will be called snd-maestro3. -config SND_MAESTRO3_FIRMWARE_IN_KERNEL - bool "In-kernel firmware for Maestro3 driver" - depends on SND_MAESTRO3 - default y - help - Say Y here to include the static firmware built in the kernel - for the Maestro3 driver. If you choose N here, you need to - install the firmware files from the alsa-firmware package. - config SND_MIXART tristate "Digigram miXart" depends on SND diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 9dfba6eff858..b2582972c9a4 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -58,10 +58,8 @@ MODULE_SUPPORTED_DEVICE("{{ESS,Maestro3 PCI}," "{ESS,Allegro PCI}," "{ESS,Allegro-1 PCI}," "{ESS,Canyon3D-2/LE PCI}}"); -#ifndef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL MODULE_FIRMWARE("ess/maestro3_assp_kernel.fw"); MODULE_FIRMWARE("ess/maestro3_assp_minisrc.fw"); -#endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -2101,161 +2099,6 @@ static int __devinit snd_m3_mixer(struct snd_m3 *chip) } -#ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL - -/* - * DSP Code images - */ - -static const u16 assp_kernel_image[] = { - 0x7980, 0x0030, 0x7980, 0x03B4, 0x7980, 0x03B4, 0x7980, 0x00FB, 0x7980, 0x00DD, 0x7980, 0x03B4, - 0x7980, 0x0332, 0x7980, 0x0287, 0x7980, 0x03B4, 0x7980, 0x03B4, 0x7980, 0x03B4, 0x7980, 0x03B4, - 0x7980, 0x031A, 0x7980, 0x03B4, 0x7980, 0x022F, 0x7980, 0x03B4, 0x7980, 0x03B4, 0x7980, 0x03B4, - 0x7980, 0x03B4, 0x7980, 0x03B4, 0x7980, 0x0063, 0x7980, 0x006B, 0x7980, 0x03B4, 0x7980, 0x03B4, - 0xBF80, 0x2C7C, 0x8806, 0x8804, 0xBE40, 0xBC20, 0xAE09, 0x1000, 0xAE0A, 0x0001, 0x6938, 0xEB08, - 0x0053, 0x695A, 0xEB08, 0x00D6, 0x0009, 0x8B88, 0x6980, 0xE388, 0x0036, 0xBE30, 0xBC20, 0x6909, - 0xB801, 0x9009, 0xBE41, 0xBE41, 0x6928, 0xEB88, 0x0078, 0xBE41, 0xBE40, 0x7980, 0x0038, 0xBE41, - 0xBE41, 0x903A, 0x6938, 0xE308, 0x0056, 0x903A, 0xBE41, 0xBE40, 0xEF00, 0x903A, 0x6939, 0xE308, - 0x005E, 0x903A, 0xEF00, 0x690B, 0x660C, 0xEF8C, 0x690A, 0x660C, 0x620B, 0x6609, 0xEF00, 0x6910, - 0x660F, 0xEF04, 0xE388, 0x0075, 0x690E, 0x660F, 0x6210, 0x660D, 0xEF00, 0x690E, 0x660D, 0xEF00, - 0xAE70, 0x0001, 0xBC20, 0xAE27, 0x0001, 0x6939, 0xEB08, 0x005D, 0x6926, 0xB801, 0x9026, 0x0026, - 0x8B88, 0x6980, 0xE388, 0x00CB, 0x9028, 0x0D28, 0x4211, 0xE100, 0x007A, 0x4711, 0xE100, 0x00A0, - 0x7A80, 0x0063, 0xB811, 0x660A, 0x6209, 0xE304, 0x007A, 0x0C0B, 0x4005, 0x100A, 0xBA01, 0x9012, - 0x0C12, 0x4002, 0x7980, 0x00AF, 0x7A80, 0x006B, 0xBE02, 0x620E, 0x660D, 0xBA10, 0xE344, 0x007A, - 0x0C10, 0x4005, 0x100E, 0xBA01, 0x9012, 0x0C12, 0x4002, 0x1003, 0xBA02, 0x9012, 0x0C12, 0x4000, - 0x1003, 0xE388, 0x00BA, 0x1004, 0x7980, 0x00BC, 0x1004, 0xBA01, 0x9012, 0x0C12, 0x4001, 0x0C05, - 0x4003, 0x0C06, 0x4004, 0x1011, 0xBFB0, 0x01FF, 0x9012, 0x0C12, 0x4006, 0xBC20, 0xEF00, 0xAE26, - 0x1028, 0x6970, 0xBFD0, 0x0001, 0x9070, 0xE388, 0x007A, 0xAE28, 0x0000, 0xEF00, 0xAE70, 0x0300, - 0x0C70, 0xB00C, 0xAE5A, 0x0000, 0xEF00, 0x7A80, 0x038A, 0x697F, 0xB801, 0x907F, 0x0056, 0x8B88, - 0x0CA0, 0xB008, 0xAF71, 0xB000, 0x4E71, 0xE200, 0x00F3, 0xAE56, 0x1057, 0x0056, 0x0CA0, 0xB008, - 0x8056, 0x7980, 0x03A1, 0x0810, 0xBFA0, 0x1059, 0xE304, 0x03A1, 0x8056, 0x7980, 0x03A1, 0x7A80, - 0x038A, 0xBF01, 0xBE43, 0xBE59, 0x907C, 0x6937, 0xE388, 0x010D, 0xBA01, 0xE308, 0x010C, 0xAE71, - 0x0004, 0x0C71, 0x5000, 0x6936, 0x9037, 0xBF0A, 0x109E, 0x8B8A, 0xAF80, 0x8014, 0x4C80, 0xBF0A, - 0x0560, 0xF500, 0xBF0A, 0x0520, 0xB900, 0xBB17, 0x90A0, 0x6917, 0xE388, 0x0148, 0x0D17, 0xE100, - 0x0127, 0xBF0C, 0x0578, 0xBF0D, 0x057C, 0x7980, 0x012B, 0xBF0C, 0x0538, 0xBF0D, 0x053C, 0x6900, - 0xE308, 0x0135, 0x8B8C, 0xBE59, 0xBB07, 0x90A0, 0xBC20, 0x7980, 0x0157, 0x030C, 0x8B8B, 0xB903, - 0x8809, 0xBEC6, 0x013E, 0x69AC, 0x90AB, 0x69AD, 0x90AB, 0x0813, 0x660A, 0xE344, 0x0144, 0x0309, - 0x830C, 0xBC20, 0x7980, 0x0157, 0x6955, 0xE388, 0x0157, 0x7C38, 0xBF0B, 0x0578, 0xF500, 0xBF0B, - 0x0538, 0xB907, 0x8809, 0xBEC6, 0x0156, 0x10AB, 0x90AA, 0x6974, 0xE388, 0x0163, 0xAE72, 0x0540, - 0xF500, 0xAE72, 0x0500, 0xAE61, 0x103B, 0x7A80, 0x02F6, 0x6978, 0xE388, 0x0182, 0x8B8C, 0xBF0C, - 0x0560, 0xE500, 0x7C40, 0x0814, 0xBA20, 0x8812, 0x733D, 0x7A80, 0x0380, 0x733E, 0x7A80, 0x0380, - 0x8B8C, 0xBF0C, 0x056C, 0xE500, 0x7C40, 0x0814, 0xBA2C, 0x8812, 0x733F, 0x7A80, 0x0380, 0x7340, - 0x7A80, 0x0380, 0x6975, 0xE388, 0x018E, 0xAE72, 0x0548, 0xF500, 0xAE72, 0x0508, 0xAE61, 0x1041, - 0x7A80, 0x02F6, 0x6979, 0xE388, 0x01AD, 0x8B8C, 0xBF0C, 0x0560, 0xE500, 0x7C40, 0x0814, 0xBA18, - 0x8812, 0x7343, 0x7A80, 0x0380, 0x7344, 0x7A80, 0x0380, 0x8B8C, 0xBF0C, 0x056C, 0xE500, 0x7C40, - 0x0814, 0xBA24, 0x8812, 0x7345, 0x7A80, 0x0380, 0x7346, 0x7A80, 0x0380, 0x6976, 0xE388, 0x01B9, - 0xAE72, 0x0558, 0xF500, 0xAE72, 0x0518, 0xAE61, 0x1047, 0x7A80, 0x02F6, 0x697A, 0xE388, 0x01D8, - 0x8B8C, 0xBF0C, 0x0560, 0xE500, 0x7C40, 0x0814, 0xBA08, 0x8812, 0x7349, 0x7A80, 0x0380, 0x734A, - 0x7A80, 0x0380, 0x8B8C, 0xBF0C, 0x056C, 0xE500, 0x7C40, 0x0814, 0xBA14, 0x8812, 0x734B, 0x7A80, - 0x0380, 0x734C, 0x7A80, 0x0380, 0xBC21, 0xAE1C, 0x1090, 0x8B8A, 0xBF0A, 0x0560, 0xE500, 0x7C40, - 0x0812, 0xB804, 0x8813, 0x8B8D, 0xBF0D, 0x056C, 0xE500, 0x7C40, 0x0815, 0xB804, 0x8811, 0x7A80, - 0x034A, 0x8B8A, 0xBF0A, 0x0560, 0xE500, 0x7C40, 0x731F, 0xB903, 0x8809, 0xBEC6, 0x01F9, 0x548A, - 0xBE03, 0x98A0, 0x7320, 0xB903, 0x8809, 0xBEC6, 0x0201, 0x548A, 0xBE03, 0x98A0, 0x1F20, 0x2F1F, - 0x9826, 0xBC20, 0x6935, 0xE388, 0x03A1, 0x6933, 0xB801, 0x9033, 0xBFA0, 0x02EE, 0xE308, 0x03A1, - 0x9033, 0xBF00, 0x6951, 0xE388, 0x021F, 0x7334, 0xBE80, 0x5760, 0xBE03, 0x9F7E, 0xBE59, 0x9034, - 0x697E, 0x0D51, 0x9013, 0xBC20, 0x695C, 0xE388, 0x03A1, 0x735E, 0xBE80, 0x5760, 0xBE03, 0x9F7E, - 0xBE59, 0x905E, 0x697E, 0x0D5C, 0x9013, 0x7980, 0x03A1, 0x7A80, 0x038A, 0xBF01, 0xBE43, 0x6977, - 0xE388, 0x024E, 0xAE61, 0x104D, 0x0061, 0x8B88, 0x6980, 0xE388, 0x024E, 0x9071, 0x0D71, 0x000B, - 0xAFA0, 0x8010, 0xAFA0, 0x8010, 0x0810, 0x660A, 0xE308, 0x0249, 0x0009, 0x0810, 0x660C, 0xE388, - 0x024E, 0x800B, 0xBC20, 0x697B, 0xE388, 0x03A1, 0xBF0A, 0x109E, 0x8B8A, 0xAF80, 0x8014, 0x4C80, - 0xE100, 0x0266, 0x697C, 0xBF90, 0x0560, 0x9072, 0x0372, 0x697C, 0xBF90, 0x0564, 0x9073, 0x0473, - 0x7980, 0x0270, 0x697C, 0xBF90, 0x0520, 0x9072, 0x0372, 0x697C, 0xBF90, 0x0524, 0x9073, 0x0473, - 0x697C, 0xB801, 0x907C, 0xBF0A, 0x10FD, 0x8B8A, 0xAF80, 0x8010, 0x734F, 0x548A, 0xBE03, 0x9880, - 0xBC21, 0x7326, 0x548B, 0xBE03, 0x618B, 0x988C, 0xBE03, 0x6180, 0x9880, 0x7980, 0x03A1, 0x7A80, - 0x038A, 0x0D28, 0x4711, 0xE100, 0x02BE, 0xAF12, 0x4006, 0x6912, 0xBFB0, 0x0C00, 0xE388, 0x02B6, - 0xBFA0, 0x0800, 0xE388, 0x02B2, 0x6912, 0xBFB0, 0x0C00, 0xBFA0, 0x0400, 0xE388, 0x02A3, 0x6909, - 0x900B, 0x7980, 0x02A5, 0xAF0B, 0x4005, 0x6901, 0x9005, 0x6902, 0x9006, 0x4311, 0xE100, 0x02ED, - 0x6911, 0xBFC0, 0x2000, 0x9011, 0x7980, 0x02ED, 0x6909, 0x900B, 0x7980, 0x02B8, 0xAF0B, 0x4005, - 0xAF05, 0x4003, 0xAF06, 0x4004, 0x7980, 0x02ED, 0xAF12, 0x4006, 0x6912, 0xBFB0, 0x0C00, 0xE388, - 0x02E7, 0xBFA0, 0x0800, 0xE388, 0x02E3, 0x6912, 0xBFB0, 0x0C00, 0xBFA0, 0x0400, 0xE388, 0x02D4, - 0x690D, 0x9010, 0x7980, 0x02D6, 0xAF10, 0x4005, 0x6901, 0x9005, 0x6902, 0x9006, 0x4311, 0xE100, - 0x02ED, 0x6911, 0xBFC0, 0x2000, 0x9011, 0x7980, 0x02ED, 0x690D, 0x9010, 0x7980, 0x02E9, 0xAF10, - 0x4005, 0xAF05, 0x4003, 0xAF06, 0x4004, 0xBC20, 0x6970, 0x9071, 0x7A80, 0x0078, 0x6971, 0x9070, - 0x7980, 0x03A1, 0xBC20, 0x0361, 0x8B8B, 0x6980, 0xEF88, 0x0272, 0x0372, 0x7804, 0x9071, 0x0D71, - 0x8B8A, 0x000B, 0xB903, 0x8809, 0xBEC6, 0x0309, 0x69A8, 0x90AB, 0x69A8, 0x90AA, 0x0810, 0x660A, - 0xE344, 0x030F, 0x0009, 0x0810, 0x660C, 0xE388, 0x0314, 0x800B, 0xBC20, 0x6961, 0xB801, 0x9061, - 0x7980, 0x02F7, 0x7A80, 0x038A, 0x5D35, 0x0001, 0x6934, 0xB801, 0x9034, 0xBF0A, 0x109E, 0x8B8A, - 0xAF80, 0x8014, 0x4880, 0xAE72, 0x0550, 0xF500, 0xAE72, 0x0510, 0xAE61, 0x1051, 0x7A80, 0x02F6, - 0x7980, 0x03A1, 0x7A80, 0x038A, 0x5D35, 0x0002, 0x695E, 0xB801, 0x905E, 0xBF0A, 0x109E, 0x8B8A, - 0xAF80, 0x8014, 0x4780, 0xAE72, 0x0558, 0xF500, 0xAE72, 0x0518, 0xAE61, 0x105C, 0x7A80, 0x02F6, - 0x7980, 0x03A1, 0x001C, 0x8B88, 0x6980, 0xEF88, 0x901D, 0x0D1D, 0x100F, 0x6610, 0xE38C, 0x0358, - 0x690E, 0x6610, 0x620F, 0x660D, 0xBA0F, 0xE301, 0x037A, 0x0410, 0x8B8A, 0xB903, 0x8809, 0xBEC6, - 0x036C, 0x6A8C, 0x61AA, 0x98AB, 0x6A8C, 0x61AB, 0x98AD, 0x6A8C, 0x61AD, 0x98A9, 0x6A8C, 0x61A9, - 0x98AA, 0x7C04, 0x8B8B, 0x7C04, 0x8B8D, 0x7C04, 0x8B89, 0x7C04, 0x0814, 0x660E, 0xE308, 0x0379, - 0x040D, 0x8410, 0xBC21, 0x691C, 0xB801, 0x901C, 0x7980, 0x034A, 0xB903, 0x8809, 0x8B8A, 0xBEC6, - 0x0388, 0x54AC, 0xBE03, 0x618C, 0x98AA, 0xEF00, 0xBC20, 0xBE46, 0x0809, 0x906B, 0x080A, 0x906C, - 0x080B, 0x906D, 0x081A, 0x9062, 0x081B, 0x9063, 0x081E, 0x9064, 0xBE59, 0x881E, 0x8065, 0x8166, - 0x8267, 0x8368, 0x8469, 0x856A, 0xEF00, 0xBC20, 0x696B, 0x8809, 0x696C, 0x880A, 0x696D, 0x880B, - 0x6962, 0x881A, 0x6963, 0x881B, 0x6964, 0x881E, 0x0065, 0x0166, 0x0267, 0x0368, 0x0469, 0x056A, - 0xBE3A, -}; - -/* - * Mini sample rate converter code image - * that is to be loaded at 0x400 on the DSP. - */ -static const u16 assp_minisrc_image[] = { - - 0xBF80, 0x101E, 0x906E, 0x006E, 0x8B88, 0x6980, 0xEF88, 0x906F, 0x0D6F, 0x6900, 0xEB08, 0x0412, - 0xBC20, 0x696E, 0xB801, 0x906E, 0x7980, 0x0403, 0xB90E, 0x8807, 0xBE43, 0xBF01, 0xBE47, 0xBE41, - 0x7A80, 0x002A, 0xBE40, 0x3029, 0xEFCC, 0xBE41, 0x7A80, 0x0028, 0xBE40, 0x3028, 0xEFCC, 0x6907, - 0xE308, 0x042A, 0x6909, 0x902C, 0x7980, 0x042C, 0x690D, 0x902C, 0x1009, 0x881A, 0x100A, 0xBA01, - 0x881B, 0x100D, 0x881C, 0x100E, 0xBA01, 0x881D, 0xBF80, 0x00ED, 0x881E, 0x050C, 0x0124, 0xB904, - 0x9027, 0x6918, 0xE308, 0x04B3, 0x902D, 0x6913, 0xBFA0, 0x7598, 0xF704, 0xAE2D, 0x00FF, 0x8B8D, - 0x6919, 0xE308, 0x0463, 0x691A, 0xE308, 0x0456, 0xB907, 0x8809, 0xBEC6, 0x0453, 0x10A9, 0x90AD, - 0x7980, 0x047C, 0xB903, 0x8809, 0xBEC6, 0x0460, 0x1889, 0x6C22, 0x90AD, 0x10A9, 0x6E23, 0x6C22, - 0x90AD, 0x7980, 0x047C, 0x101A, 0xE308, 0x046F, 0xB903, 0x8809, 0xBEC6, 0x046C, 0x10A9, 0x90A0, - 0x90AD, 0x7980, 0x047C, 0xB901, 0x8809, 0xBEC6, 0x047B, 0x1889, 0x6C22, 0x90A0, 0x90AD, 0x10A9, - 0x6E23, 0x6C22, 0x90A0, 0x90AD, 0x692D, 0xE308, 0x049C, 0x0124, 0xB703, 0xB902, 0x8818, 0x8B89, - 0x022C, 0x108A, 0x7C04, 0x90A0, 0x692B, 0x881F, 0x7E80, 0x055B, 0x692A, 0x8809, 0x8B89, 0x99A0, - 0x108A, 0x90A0, 0x692B, 0x881F, 0x7E80, 0x055B, 0x692A, 0x8809, 0x8B89, 0x99AF, 0x7B99, 0x0484, - 0x0124, 0x060F, 0x101B, 0x2013, 0x901B, 0xBFA0, 0x7FFF, 0xE344, 0x04AC, 0x901B, 0x8B89, 0x7A80, - 0x051A, 0x6927, 0xBA01, 0x9027, 0x7A80, 0x0523, 0x6927, 0xE308, 0x049E, 0x7980, 0x050F, 0x0624, - 0x1026, 0x2013, 0x9026, 0xBFA0, 0x7FFF, 0xE304, 0x04C0, 0x8B8D, 0x7A80, 0x051A, 0x7980, 0x04B4, - 0x9026, 0x1013, 0x3026, 0x901B, 0x8B8D, 0x7A80, 0x051A, 0x7A80, 0x0523, 0x1027, 0xBA01, 0x9027, - 0xE308, 0x04B4, 0x0124, 0x060F, 0x8B89, 0x691A, 0xE308, 0x04EA, 0x6919, 0xE388, 0x04E0, 0xB903, - 0x8809, 0xBEC6, 0x04DD, 0x1FA0, 0x2FAE, 0x98A9, 0x7980, 0x050F, 0xB901, 0x8818, 0xB907, 0x8809, - 0xBEC6, 0x04E7, 0x10EE, 0x90A9, 0x7980, 0x050F, 0x6919, 0xE308, 0x04FE, 0xB903, 0x8809, 0xBE46, - 0xBEC6, 0x04FA, 0x17A0, 0xBE1E, 0x1FAE, 0xBFBF, 0xFF00, 0xBE13, 0xBFDF, 0x8080, 0x99A9, 0xBE47, - 0x7980, 0x050F, 0xB901, 0x8809, 0xBEC6, 0x050E, 0x16A0, 0x26A0, 0xBFB7, 0xFF00, 0xBE1E, 0x1EA0, - 0x2EAE, 0xBFBF, 0xFF00, 0xBE13, 0xBFDF, 0x8080, 0x99A9, 0x850C, 0x860F, 0x6907, 0xE388, 0x0516, - 0x0D07, 0x8510, 0xBE59, 0x881E, 0xBE4A, 0xEF00, 0x101E, 0x901C, 0x101F, 0x901D, 0x10A0, 0x901E, - 0x10A0, 0x901F, 0xEF00, 0x101E, 0x301C, 0x9020, 0x731B, 0x5420, 0xBE03, 0x9825, 0x1025, 0x201C, - 0x9025, 0x7325, 0x5414, 0xBE03, 0x8B8E, 0x9880, 0x692F, 0xE388, 0x0539, 0xBE59, 0xBB07, 0x6180, - 0x9880, 0x8BA0, 0x101F, 0x301D, 0x9021, 0x731B, 0x5421, 0xBE03, 0x982E, 0x102E, 0x201D, 0x902E, - 0x732E, 0x5415, 0xBE03, 0x9880, 0x692F, 0xE388, 0x054F, 0xBE59, 0xBB07, 0x6180, 0x9880, 0x8BA0, - 0x6918, 0xEF08, 0x7325, 0x5416, 0xBE03, 0x98A0, 0x732E, 0x5417, 0xBE03, 0x98A0, 0xEF00, 0x8BA0, - 0xBEC6, 0x056B, 0xBE59, 0xBB04, 0xAA90, 0xBE04, 0xBE1E, 0x99E0, 0x8BE0, 0x69A0, 0x90D0, 0x69A0, - 0x90D0, 0x081F, 0xB805, 0x881F, 0x8B90, 0x69A0, 0x90D0, 0x69A0, 0x9090, 0x8BD0, 0x8BD8, 0xBE1F, - 0xEF00, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, -}; - -static const struct firmware assp_kernel = { - .data = (u8 *)assp_kernel_image, - .size = sizeof assp_kernel_image -}; -static const struct firmware assp_minisrc = { - .data = (u8 *)assp_minisrc_image, - .size = sizeof assp_minisrc_image -}; - -#ifdef __LITTLE_ENDIAN -static inline void snd_m3_convert_to_le(const struct firmware *fw) { } -#else -static void snd_m3_convert_to_le(const struct firmware *fw) -{ - int i; - u16 *data = (u16 *)fw->data; - - for (i = 0; i < fw->size / 2; ++i) - cpu_to_le16s(&data[i]); -} -#endif - -#endif /* CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL */ - - /* * initialize ASSP */ @@ -2547,10 +2390,8 @@ static int snd_m3_free(struct snd_m3 *chip) if (chip->iobase) pci_release_regions(chip->pci); -#ifndef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL release_firmware(chip->assp_kernel_image); release_firmware(chip->assp_minisrc_image); -#endif pci_disable_device(chip->pci); kfree(chip); @@ -2740,27 +2581,19 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, return -ENOMEM; } -#ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL - chip->assp_kernel_image = &assp_kernel; -#else err = request_firmware(&chip->assp_kernel_image, "ess/maestro3_assp_kernel.fw", &pci->dev); if (err < 0) { snd_m3_free(chip); return err; } -#endif -#ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL - chip->assp_minisrc_image = &assp_minisrc; -#else err = request_firmware(&chip->assp_minisrc_image, "ess/maestro3_assp_minisrc.fw", &pci->dev); if (err < 0) { snd_m3_free(chip); return err; } -#endif if ((err = pci_request_regions(pci, card->driver)) < 0) { snd_m3_free(chip); @@ -2912,10 +2745,6 @@ static struct pci_driver driver = { static int __init alsa_card_m3_init(void) { -#ifdef CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL - snd_m3_convert_to_le(&assp_kernel); - snd_m3_convert_to_le(&assp_minisrc); -#endif return pci_register_driver(&driver); } -- cgit v1.2.3 From 18ee6dfae89d9c131e3c9952939633ba8fa86247 Mon Sep 17 00:00:00 2001 From: David Woodhouse Date: Thu, 29 May 2008 15:07:34 +0300 Subject: firmware: convert ymfpci driver to use firmware loader exclusively Signed-off-by: David Woodhouse --- sound/pci/Kconfig | 10 - sound/pci/ymfpci/ymfpci_image.h | 1565 --------------------------------------- sound/pci/ymfpci/ymfpci_main.c | 63 -- 3 files changed, 1638 deletions(-) delete mode 100644 sound/pci/ymfpci/ymfpci_image.h (limited to 'sound/pci') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 32836ea45170..e4a0045e2a31 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -914,7 +914,6 @@ config SND_VX222 config SND_YMFPCI tristate "Yamaha YMF724/740/744/754" depends on SND - select FW_LOADER if !SND_YMFPCI_FIRMWARE_IN_KERNEL select SND_OPL3_LIB select SND_MPU401_UART select SND_AC97_CODEC @@ -925,15 +924,6 @@ config SND_YMFPCI To compile this driver as a module, choose M here: the module will be called snd-ymfpci. -config SND_YMFPCI_FIRMWARE_IN_KERNEL - bool "In-kernel firmware for YMFPCI driver" - depends on SND_YMFPCI - default y - help - Say Y here to include the static firmware built in the kernel - for the YMFPCI driver. If you choose N here, you need to - install the firmware files from the alsa-firmware package. - config SND_AC97_POWER_SAVE bool "AC97 Power-Saving Mode" depends on SND_AC97_CODEC && EXPERIMENTAL diff --git a/sound/pci/ymfpci/ymfpci_image.h b/sound/pci/ymfpci/ymfpci_image.h deleted file mode 100644 index 112f2fff6c8e..000000000000 --- a/sound/pci/ymfpci/ymfpci_image.h +++ /dev/null @@ -1,1565 +0,0 @@ -#ifndef _HWMCODE_ -#define _HWMCODE_ - -static u32 DspInst[YDSXG_DSPLENGTH / 4] = { - 0x00000081, 0x000001a4, 0x0000000a, 0x0000002f, - 0x00080253, 0x01800317, 0x0000407b, 0x0000843f, - 0x0001483c, 0x0001943c, 0x0005d83c, 0x00001c3c, - 0x0000c07b, 0x00050c3f, 0x0121503c, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, 0x00000000 -}; - -static u32 CntrlInst[YDSXG_CTRLLENGTH / 4] = { - 0x000007, 0x240007, 0x0C0007, 0x1C0007, - 0x060007, 0x700002, 0x000020, 0x030040, - 0x007104, 0x004286, 0x030040, 0x000F0D, - 0x000810, 0x20043A, 0x000282, 0x00020D, - 0x000810, 0x20043A, 0x001282, 0x200E82, - 0x001A82, 0x032D0D, 0x000810, 0x10043A, - 0x02D38D, 0x000810, 0x18043A, 0x00010D, - 0x020015, 0x0000FD, 0x000020, 0x038860, - 0x039060, 0x038060, 0x038040, 0x038040, - 0x038040, 0x018040, 0x000A7D, 0x038040, - 0x038040, 0x018040, 0x200402, 0x000882, - 0x08001A, 0x000904, 0x015986, 0x000007, - 0x260007, 0x000007, 0x000007, 0x018A06, - 0x000007, 0x030C8D, 0x000810, 0x18043A, - 0x260007, 0x00087D, 0x018042, 0x00160A, - 0x04A206, 0x000007, 0x00218D, 0x000810, - 0x08043A, 0x21C206, 0x000007, 0x0007FD, - 0x018042, 0x08000A, 0x000904, 0x029386, - 0x000195, 0x090D04, 0x000007, 0x000820, - 0x0000F5, 0x000B7D, 0x01F060, 0x0000FD, - 0x032206, 0x018040, 0x000A7D, 0x038042, - 0x13804A, 0x18000A, 0x001820, 0x059060, - 0x058860, 0x018040, 0x0000FD, 0x018042, - 0x70000A, 0x000115, 0x071144, 0x032386, - 0x030000, 0x007020, 0x034A06, 0x018040, - 0x00348D, 0x000810, 0x08043A, 0x21EA06, - 0x000007, 0x02D38D, 0x000810, 0x18043A, - 0x018206, 0x000007, 0x240007, 0x000F8D, - 0x000810, 0x00163A, 0x002402, 0x005C02, - 0x0028FD, 0x000020, 0x018040, 0x08000D, - 0x000815, 0x510984, 0x000007, 0x00004D, - 0x000E5D, 0x000E02, 0x00418D, 0x000810, - 0x08043A, 0x2C8A06, 0x000007, 0x00008D, - 0x000924, 0x000F02, 0x00458D, 0x000810, - 0x08043A, 0x2C8A06, 0x000007, 0x00387D, - 0x018042, 0x08000A, 0x001015, 0x010984, - 0x018386, 0x000007, 0x01AA06, 0x000007, - 0x0008FD, 0x018042, 0x18000A, 0x001904, - 0x218086, 0x280007, 0x001810, 0x28043A, - 0x280C02, 0x00000D, 0x000810, 0x28143A, - 0x08808D, 0x000820, 0x0002FD, 0x018040, - 0x200007, 0x00020D, 0x189904, 0x000007, - 0x00402D, 0x0000BD, 0x0002FD, 0x018042, - 0x08000A, 0x000904, 0x055A86, 0x000007, - 0x000100, 0x000A20, 0x00047D, 0x018040, - 0x018042, 0x20000A, 0x003015, 0x012144, - 0x034986, 0x000007, 0x002104, 0x034986, - 0x000007, 0x000F8D, 0x000810, 0x280C3A, - 0x023944, 0x06C986, 0x000007, 0x001810, - 0x28043A, 0x08810D, 0x000820, 0x0002FD, - 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0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000, - 0x000000, 0x000000, 0x000000, 0x000000 -}; - -// -------------------------------------------- -// DS-1E Controller InstructionRAM Code -// 1999/06/21 -// Buf441 slot is Enabled. -// -------------------------------------------- -// 04/09 creat -// 04/12 stop nise fix -// 06/21 WorkingOff timming -static u32 CntrlInst1E[YDSXG_CTRLLENGTH / 4] = { - 0x000007, 0x240007, 0x0C0007, 0x1C0007, - 0x060007, 0x700002, 0x000020, 0x030040, - 0x007104, 0x004286, 0x030040, 0x000F0D, - 0x000810, 0x20043A, 0x000282, 0x00020D, - 0x000810, 0x20043A, 0x001282, 0x200E82, - 0x00800D, 0x000810, 0x20043A, 0x001A82, - 0x03460D, 0x000810, 0x10043A, 0x02EC0D, - 0x000810, 0x18043A, 0x00010D, 0x020015, - 0x0000FD, 0x000020, 0x038860, 0x039060, - 0x038060, 0x038040, 0x038040, 0x038040, - 0x018040, 0x000A7D, 0x038040, 0x038040, - 0x018040, 0x200402, 0x000882, 0x08001A, - 0x000904, 0x017186, 0x000007, 0x260007, - 0x400007, 0x000007, 0x03258D, 0x000810, - 0x18043A, 0x260007, 0x284402, 0x00087D, - 0x018042, 0x00160A, 0x05A206, 0x000007, - 0x440007, 0x00230D, 0x000810, 0x08043A, - 0x22FA06, 0x000007, 0x0007FD, 0x018042, - 0x08000A, 0x000904, 0x02AB86, 0x000195, - 0x090D04, 0x000007, 0x000820, 0x0000F5, - 0x000B7D, 0x01F060, 0x0000FD, 0x033A06, - 0x018040, 0x000A7D, 0x038042, 0x13804A, - 0x18000A, 0x001820, 0x059060, 0x058860, - 0x018040, 0x0000FD, 0x018042, 0x70000A, - 0x000115, 0x071144, 0x033B86, 0x030000, - 0x007020, 0x036206, 0x018040, 0x00360D, - 0x000810, 0x08043A, 0x232206, 0x000007, - 0x02EC0D, 0x000810, 0x18043A, 0x019A06, - 0x000007, 0x240007, 0x000F8D, 0x000810, - 0x00163A, 0x002402, 0x005C02, 0x0028FD, - 0x000020, 0x018040, 0x08000D, 0x000815, - 0x510984, 0x000007, 0x00004D, 0x000E5D, - 0x000E02, 0x00430D, 0x000810, 0x08043A, - 0x2E1206, 0x000007, 0x00008D, 0x000924, - 0x000F02, 0x00470D, 0x000810, 0x08043A, - 0x2E1206, 0x000007, 0x480480, 0x001210, - 0x28043A, 0x00778D, 0x000810, 0x280C3A, - 0x00068D, 0x000810, 0x28143A, 0x284402, - 0x03258D, 0x000810, 0x18043A, 0x07FF8D, - 0x000820, 0x0002FD, 0x018040, 0x260007, - 0x200007, 0x0002FD, 0x018042, 0x08000A, - 0x000904, 0x051286, 0x000007, 0x240007, - 0x02EC0D, 0x000810, 0x18043A, 0x00387D, - 0x018042, 0x08000A, 0x001015, 0x010984, - 0x019B86, 0x000007, 0x01B206, 0x000007, - 0x0008FD, 0x018042, 0x18000A, 0x001904, - 0x22B886, 0x280007, 0x001810, 0x28043A, - 0x280C02, 0x00000D, 0x000810, 0x28143A, - 0x08808D, 0x000820, 0x0002FD, 0x018040, - 0x200007, 0x00020D, 0x189904, 0x000007, - 0x00402D, 0x0000BD, 0x0002FD, 0x018042, - 0x08000A, 0x000904, 0x065A86, 0x000007, - 0x000100, 0x000A20, 0x00047D, 0x018040, - 0x018042, 0x20000A, 0x003015, 0x012144, - 0x036186, 0x000007, 0x002104, 0x036186, - 0x000007, 0x000F8D, 0x000810, 0x280C3A, - 0x023944, 0x07C986, 0x000007, 0x001810, - 0x28043A, 0x08810D, 0x000820, 0x0002FD, - 0x018040, 0x200007, 0x002810, 0x78003A, - 0x00788D, 0x000810, 0x08043A, 0x2A1206, - 0x000007, 0x00400D, 0x001015, 0x189904, - 0x292904, 0x393904, 0x000007, 0x070206, - 0x000007, 0x0004F5, 0x00007D, 0x000020, - 0x00008D, 0x010860, 0x018040, 0x00047D, - 0x038042, 0x21804A, 0x18000A, 0x021944, - 0x229086, 0x000007, 0x004075, 0x71F104, - 0x000007, 0x010042, 0x28000A, 0x002904, - 0x225886, 0x000007, 0x003C0D, 0x30A904, - 0x000007, 0x00077D, 0x018042, 0x08000A, - 0x000904, 0x08DA86, 0x00057D, 0x002820, - 0x03B060, 0x08F206, 0x018040, 0x003020, - 0x03A860, 0x018040, 0x0002FD, 0x018042, - 0x08000A, 0x000904, 0x08FA86, 0x000007, - 0x00057D, 0x018042, 0x28040A, 0x000E8D, - 0x000810, 0x280C3A, 0x00000D, 0x000810, - 0x28143A, 0x09000D, 0x000820, 0x0002FD, - 0x018040, 0x200007, 0x003DFD, 0x000020, - 0x018040, 0x00107D, 0x009D8D, 0x000810, - 0x08043A, 0x2A1206, 0x000007, 0x000815, - 0x08001A, 0x010984, 0x0A5186, 0x00137D, - 0x200500, 0x280F20, 0x338F60, 0x3B8F60, - 0x438F60, 0x4B8F60, 0x538F60, 0x5B8F60, - 0x038A60, 0x018040, 0x00107D, 0x018042, - 0x08000A, 0x000215, 0x010984, 0x3A8186, - 0x000007, 0x007FBD, 0x383DC4, 0x000007, - 0x001A7D, 0x001375, 0x018042, 0x09004A, - 0x10000A, 0x0B8D04, 0x139504, 0x000007, - 0x000820, 0x019060, 0x001104, 0x225886, - 0x010040, 0x0017FD, 0x018042, 0x08000A, - 0x000904, 0x225A86, 0x000007, 0x00197D, - 0x038042, 0x09804A, 0x10000A, 0x000924, - 0x001664, 0x0011FD, 0x038042, 0x2B804A, - 0x19804A, 0x00008D, 0x218944, 0x000007, - 0x002244, 0x0C1986, 0x000007, 0x001A64, - 0x002A24, 0x00197D, 0x080102, 0x100122, - 0x000820, 0x039060, 0x018040, 0x003DFD, - 0x00008D, 0x000820, 0x018040, 0x001375, - 0x001A7D, 0x010042, 0x09804A, 0x10000A, - 0x00021D, 0x0189E4, 0x2992E4, 0x309144, - 0x000007, 0x00060D, 0x000A15, 0x000C1D, - 0x001025, 0x00A9E4, 0x012BE4, 0x000464, - 0x01B3E4, 0x0232E4, 0x000464, 0x000464, - 0x000464, 0x000464, 0x00040D, 0x08B1C4, - 0x000007, 0x000820, 0x000BF5, 0x030040, - 0x00197D, 0x038042, 0x09804A, 0x000A24, - 0x08000A, 0x080E64, 0x000007, 0x100122, - 0x000820, 0x031060, 0x010040, 0x0064AC, - 0x00027D, 0x000020, 0x018040, 0x00107D, - 0x018042, 0x0011FD, 0x3B804A, 0x09804A, - 0x20000A, 0x000095, 0x1A1144, 0x00A144, - 0x0E5886, 0x00040D, 0x00B984, 0x0E5986, - 0x0018FD, 0x018042, 0x0010FD, 0x09804A, - 0x28000A, 0x000095, 0x010924, 0x002A64, - 0x0E4986, 0x000007, 0x002904, 0x0E5A86, - 0x000007, 0x0E6206, 0x080002, 0x00008D, - 0x00387D, 0x000820, 0x018040, 0x00127D, - 0x018042, 0x10000A, 0x003904, 0x0F0986, - 0x00080D, 0x7FFFB5, 0x00B984, 0x0ED986, - 0x000025, 0x0FB206, 0x00002D, 0x000015, - 0x00082D, 0x02E00D, 0x000820, 0x0FFA06, - 0x00000D, 0x7F8035, 0x00B984, 0x0FA986, - 0x400025, 0x00008D, 0x110944, 0x000007, - 0x00018D, 0x109504, 0x000007, 0x009164, - 0x000424, 0x000424, 0x000424, 0x100102, - 0x280002, 0x02DF0D, 0x000820, 0x0FFA06, - 0x00018D, 0x00042D, 0x00008D, 0x109504, - 0x000007, 0x00020D, 0x109184, 0x000007, - 0x02DF8D, 0x000820, 0x00008D, 0x0038FD, - 0x018040, 0x003BFD, 0x001020, 0x03A860, - 0x000815, 0x313184, 0x212184, 0x000007, - 0x03B060, 0x03A060, 0x018040, 0x0022FD, - 0x000095, 0x010924, 0x000424, 0x000424, - 0x001264, 0x100102, 0x000820, 0x039060, - 0x018040, 0x001924, 0x010F0D, 0x00397D, - 0x000820, 0x058040, 0x038042, 0x09844A, - 0x000606, 0x08040A, 0x000424, 0x000424, - 0x00117D, 0x018042, 0x08000A, 0x000A24, - 0x280502, 0x280C02, 0x09800D, 0x000820, - 0x0002FD, 0x018040, 0x200007, 0x0022FD, - 0x018042, 0x08000A, 0x000095, 0x280DC4, - 0x011924, 0x00197D, 0x018042, 0x0011FD, - 0x09804A, 0x10000A, 0x0000B5, 0x113144, - 0x0A8D04, 0x000007, 0x080A44, 0x129504, - 0x000007, 0x0023FD, 0x001020, 0x038040, - 0x101244, 0x000007, 0x000820, 0x039060, - 0x018040, 0x0002FD, 0x018042, 0x08000A, - 0x000904, 0x123286, 0x000007, 0x003BFD, - 0x000100, 0x000A10, 0x0B807A, 0x13804A, - 0x090984, 0x000007, 0x000095, 0x013D04, - 0x12B886, 0x10000A, 0x100002, 0x090984, - 0x000007, 0x038042, 0x11804A, 0x090D04, - 0x000007, 0x10000A, 0x090D84, 0x000007, - 0x00257D, 0x000820, 0x018040, 0x00010D, - 0x000810, 0x28143A, 0x00127D, 0x018042, - 0x20000A, 0x00197D, 0x018042, 0x00117D, - 0x31804A, 0x10000A, 0x003124, 0x013B8D, - 0x00397D, 0x000820, 0x058040, 0x038042, - 0x09844A, 0x000606, 0x08040A, 0x300102, - 0x003124, 0x000424, 0x000424, 0x001224, - 0x280502, 0x001A4C, 0x143986, 0x700002, - 0x00002D, 0x030000, 0x00387D, 0x018042, - 0x10000A, 0x146206, 0x002124, 0x0000AD, - 0x100002, 0x00010D, 0x000924, 0x006B24, - 0x014A0D, 0x00397D, 0x000820, 0x058040, - 0x038042, 0x09844A, 0x000606, 0x08040A, - 0x003264, 0x00008D, 0x000A24, 0x001020, - 0x00227D, 0x018040, 0x014F8D, 0x000810, - 0x08043A, 0x2B5A06, 0x000007, 0x002820, - 0x00207D, 0x018040, 0x00117D, 0x038042, - 0x13804A, 0x33800A, 0x00387D, 0x018042, - 0x08000A, 0x000904, 0x177286, 0x000007, - 0x00008D, 0x030964, 0x015B0D, 0x00397D, - 0x000820, 0x058040, 0x038042, 0x09844A, - 0x000606, 0x08040A, 0x380102, 0x000424, - 0x000424, 0x001224, 0x0002FD, 0x018042, - 0x08000A, 0x000904, 0x15DA86, 0x000007, - 0x280502, 0x001A4C, 0x177186, 0x000007, - 0x032164, 0x00632C, 0x003DFD, 0x018042, - 0x08000A, 0x000095, 0x090904, 0x000007, - 0x000820, 0x001A4C, 0x169986, 0x018040, - 0x030000, 0x16B206, 0x002124, 0x00010D, - 0x000924, 0x006B24, 0x016F0D, 0x00397D, - 0x000820, 0x058040, 0x038042, 0x09844A, - 0x000606, 0x08040A, 0x003A64, 0x000095, - 0x001224, 0x0002FD, 0x018042, 0x08000A, - 0x000904, 0x171286, 0x000007, 0x01760D, - 0x000810, 0x08043A, 0x2B5A06, 0x000007, - 0x160A06, 0x000007, 0x007020, 0x08010A, - 0x10012A, 0x0020FD, 0x038860, 0x039060, - 0x018040, 0x00227D, 0x018042, 0x003DFD, - 0x08000A, 0x31844A, 0x000904, 0x181086, - 0x18008B, 0x00008D, 0x189904, 0x00312C, - 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-#endif //_HWMCODE_ diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 6298b29c66bb..2164e34e46b7 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1994,65 +1994,6 @@ static void snd_ymfpci_disable_dsp(struct snd_ymfpci *chip) } } -#ifdef CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL - -#include "ymfpci_image.h" - -static struct firmware snd_ymfpci_dsp_microcode = { - .size = YDSXG_DSPLENGTH, - .data = (u8 *)DspInst, -}; -static struct firmware snd_ymfpci_controller_microcode = { - .size = YDSXG_CTRLLENGTH, - .data = (u8 *)CntrlInst, -}; -static struct firmware snd_ymfpci_controller_1e_microcode = { - .size = YDSXG_CTRLLENGTH, - .data = (u8 *)CntrlInst1E, -}; - -#ifdef __BIG_ENDIAN -static int microcode_swapped; -static DEFINE_MUTEX(microcode_swap); - -static void snd_ymfpci_convert_to_le(const struct firmware *fw) -{ - int i; - u32 *data = (u32 *)fw->data; - - for (i = 0; i < fw->size / 4; ++i) - cpu_to_le32s(&data[i]); -} -#endif - -static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) -{ -#ifdef __BIG_ENDIAN - mutex_lock(µcode_swap); - if (!microcode_swapped) { - snd_ymfpci_convert_to_le(&snd_ymfpci_dsp_microcode); - snd_ymfpci_convert_to_le(&snd_ymfpci_controller_1e_microcode); - snd_ymfpci_convert_to_le(&snd_ymfpci_controller_microcode); - microcode_swapped = 1; - } - mutex_unlock(µcode_swap); -#endif - - chip->dsp_microcode = &snd_ymfpci_dsp_microcode; - if (chip->device_id == PCI_DEVICE_ID_YAMAHA_724F || - chip->device_id == PCI_DEVICE_ID_YAMAHA_740C || - chip->device_id == PCI_DEVICE_ID_YAMAHA_744 || - chip->device_id == PCI_DEVICE_ID_YAMAHA_754) - chip->controller_microcode = - &snd_ymfpci_controller_1e_microcode; - else - chip->controller_microcode = - &snd_ymfpci_controller_microcode; - return 0; -} - -#else /* use fw_loader */ - static int snd_ymfpci_request_firmware(struct snd_ymfpci *chip) { int err, is_1e; @@ -2091,8 +2032,6 @@ MODULE_FIRMWARE("yamaha/ds1_dsp.fw"); MODULE_FIRMWARE("yamaha/ds1_ctrl.fw"); MODULE_FIRMWARE("yamaha/ds1e_ctrl.fw"); -#endif - static void snd_ymfpci_download_image(struct snd_ymfpci *chip) { int i; @@ -2273,10 +2212,8 @@ static int snd_ymfpci_free(struct snd_ymfpci *chip) pci_write_config_word(chip->pci, 0x40, chip->old_legacy_ctrl); pci_disable_device(chip->pci); -#ifndef CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL release_firmware(chip->dsp_microcode); release_firmware(chip->controller_microcode); -#endif kfree(chip); return 0; } -- cgit v1.2.3 From f7c5dda23a9f4b7f8977612154daef44cc0f423b Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 10 Jul 2008 17:49:11 +0200 Subject: ALSA: hda: 92hd71bxx PC Beep Added volume controls for the analog PC Beep on 92hd71bxx codecs. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a6d138831e26..08cb77f51880 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -825,6 +825,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ -- cgit v1.2.3 From 4090dffb1438e03a434e3747b14321440561d956 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:02:45 +0200 Subject: ALSA: hda - Fix internal mic vref pin setup Set the vref80 to the internal mic pin 0x12 for Cx5045. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 95e3367d8879..71c4072c9266 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -686,7 +686,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, /* HP, Amp */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, -- cgit v1.2.3 From 86376df6ad0f1a7a1118cd53b0cfd679524f5436 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:04:05 +0200 Subject: ALSA: hda - Fix missing init for unsol events on micsense model Fixed the missing initialization for unsolicited events on Cx5045 micsense model. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71c4072c9266..8295994665e7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -962,6 +962,7 @@ static int patch_cxt5045(struct hda_codec *codec) codec->patch_ops.init = cxt5045_init; break; case CXT5045_LAPTOP_MICSENSE: + codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; -- cgit v1.2.3 From 9e4641541e9681a568483133813332cfafa34d86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 12 Jul 2008 12:05:25 +0200 Subject: ALSA: hda - Fix FSC V5505 model model=laptop-hpmicsense matches better to FSC V5505 laptop. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8295994665e7..7c1eb23f0cec 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -910,7 +910,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK(0x1734, 0x110e, "Fujitsu V5505", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), -- cgit v1.2.3 From 80ca9a706b458d09b8cc8d5258bb61957f66ca5e Mon Sep 17 00:00:00 2001 From: Milton Miller Date: Sun, 13 Jul 2008 13:58:12 +0200 Subject: ALSA: correct kcalloc usage kcalloc is supposed to be called with the count as its first argument and the element size as the second. Both arguments are size_t so does not affect correctness. This callsite is during module_init and therefore not performance critical. Another patch will optimize the case when the count is variable but the size is fixed. Signed-off-by: Milton Miller Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/nm256/nm256.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7efb838d18a6..06d13e717114 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1302,8 +1302,8 @@ snd_nm256_mixer(struct nm256 *chip) .read = snd_nm256_ac97_read, }; - chip->ac97_regs = kcalloc(sizeof(short), - ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL); + chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val), + sizeof(short), GFP_KERNEL); if (! chip->ac97_regs) return -ENOMEM; -- cgit v1.2.3