diff options
Diffstat (limited to 'sound')
27 files changed, 1454 insertions, 103 deletions
diff --git a/sound/core/info.c b/sound/core/info.c index 9977ec2eace3..cb5ead3e202d 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -544,7 +544,7 @@ int __init snd_info_init(void) { struct proc_dir_entry *p; - p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, &proc_root); + p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, NULL); if (p == NULL) return -ENOMEM; snd_proc_root = p; @@ -594,7 +594,7 @@ int __exit snd_info_done(void) #ifdef CONFIG_SND_OSSEMUL snd_info_free_entry(snd_oss_root); #endif - snd_remove_proc_entry(&proc_root, snd_proc_root); + snd_remove_proc_entry(NULL, snd_proc_root); } return 0; } diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 920e5780c228..23b7bc02728b 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -629,9 +629,8 @@ static const struct file_operations snd_mem_proc_fops = { static int __init snd_mem_init(void) { #ifdef CONFIG_PROC_FS - snd_mem_proc = create_proc_entry(SND_MEM_PROC_FILE, 0644, NULL); - if (snd_mem_proc) - snd_mem_proc->proc_fops = &snd_mem_proc_fops; + snd_mem_proc = proc_create(SND_MEM_PROC_FILE, 0644, NULL, + &snd_mem_proc_fops); #endif return 0; } diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index fe85af1c5693..a78a8d045175 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -8,6 +8,8 @@ config SND_PCSP tristate "Internal PC speaker support" depends on X86_PC && HIGH_RES_TIMERS depends on INPUT + depends on SND + select SND_PCM help If you don't have a sound card in your computer, you can include a driver for the PC speaker which allows it to act like a primitive diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 18cca2457d44..2af09996a3d0 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -243,7 +243,7 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, #endif } mpu->write(mpu, cmd, MPU401C(mpu)); - if (ack) { + if (ack && !(mpu->info_flags & MPU401_INFO_NO_ACK)) { ok = 0; timeout = 10000; while (!ok && timeout-- > 0) { diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 59203511e77d..54a1f9036c66 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -194,6 +194,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip) spin_unlock_irq(&chip->substream_lock); } +#ifdef CONFIG_PM static int pcsp_suspend(struct platform_device *dev, pm_message_t state) { struct snd_pcsp *chip = platform_get_drvdata(dev); @@ -201,6 +202,9 @@ static int pcsp_suspend(struct platform_device *dev, pm_message_t state) snd_pcm_suspend_all(chip->pcm); return 0; } +#else +#define pcsp_suspend NULL +#endif /* CONFIG_PM */ static void pcsp_shutdown(struct platform_device *dev) { diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 581debf37dcb..7e4742109572 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -515,19 +515,16 @@ config SND_FM801 config SND_FM801_TEA575X_BOOL bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 + depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801 help Say Y here to include support for soundcards based on the ForteMedia FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media Forte SF256-PCS-02) into the snd-fm801 driver. - This will enable support for the old V4L1 API. - config SND_FM801_TEA575X tristate depends on SND_FM801_TEA575X_BOOL default SND_FM801 - select VIDEO_V4L1 - select VIDEO_DEV config SND_HDA_INTEL tristate "Intel HD Audio" diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 39198e505b12..2da89810ca10 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3446,6 +3446,7 @@ static const struct snd_kcontrol_new snd_ac97_controls_vt1617a[] = { int patch_vt1617a(struct snd_ac97 * ac97) { int err = 0; + int val; /* we choose to not fail out at this point, but we tell the caller when we return */ @@ -3456,7 +3457,13 @@ int patch_vt1617a(struct snd_ac97 * ac97) /* bring analog power consumption to normal by turning off the * headphone amplifier, like WinXP driver for EPIA SP */ - snd_ac97_write_cache(ac97, 0x5c, 0x20); + /* We need to check the bit before writing it. + * On some (many?) hardwares, setting bit actually clears it! + */ + val = snd_ac97_read(ac97, 0x5c); + if (!(val & 0x20)) + snd_ac97_write_cache(ac97, 0x5c, 0x20); + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; ac97->build_ops = &patch_vt1616_ops; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cdda64b02f46..6d4df45e81e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -60,6 +60,7 @@ enum { ALC880_TCL_S700, ALC880_LG, ALC880_LG_LW, + ALC880_MEDION_RIM, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -2275,6 +2276,75 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res) alc880_lg_lw_automute(codec); } +static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Internal Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct hda_input_mux alc880_medion_rim_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + }, +}; + +static struct hda_verb alc880_medion_rim_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Mic1 (rear panel) pin widget for input and vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mic2 (as headphone out) for HP output */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Internal Speaker */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, + + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_medion_rim_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + if (present) + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); + else + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); +} + +static void alc880_medion_rim_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + if ((res >> 28) == ALC880_HP_EVENT) + alc880_medion_rim_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE static struct hda_amp_list alc880_loopbacks[] = { { 0x0b, HDA_INPUT, 0 }, @@ -2882,6 +2952,7 @@ static const char *alc880_models[ALC880_MODEL_LAST] = { [ALC880_F1734] = "F1734", [ALC880_LG] = "lg", [ALC880_LG_LW] = "lg-lw", + [ALC880_MEDION_RIM] = "medion", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -2933,6 +3004,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), + SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_MEDION_RIM), 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), @@ -3227,6 +3299,20 @@ static struct alc_config_preset alc880_presets[] = { .unsol_event = alc880_lg_lw_unsol_event, .init_hook = alc880_lg_lw_automute, }, + [ALC880_MEDION_RIM] = { + .mixers = { alc880_medion_rim_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_medion_rim_init_verbs, + alc_gpio2_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), + .dac_nids = alc880_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), + .channel_mode = alc880_2_jack_modes, + .input_mux = &alc880_medion_rim_capture_source, + .unsol_event = alc880_medion_rim_unsol_event, + .init_hook = alc880_medion_rim_automute, + }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, @@ -11816,7 +11902,10 @@ static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { - alc_set_pin_output(codec, nid, pin_type); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_type); + snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); } static void alc861_auto_init_multi_out(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b3a15d616873..393f7fd2b1be 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4289,6 +4289,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x }, { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x }, { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x }, + { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x }, + { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x }, /* The following does not take into account .id=0x83847661 when subsys = * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are * currently not fully supported. diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 4490422fb930..67350901772c 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2429,6 +2429,7 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, 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) { @@ -2442,12 +2443,10 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, outb(inb(ICEREG1724(ice, IRQMASK)) & ~(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX), ICEREG1724(ice, IRQMASK)); -#if 0 /* for testing */ /* set watermarks */ outb(VT1724_MPU_RX_FIFO | 0x1, ICEREG1724(ice, MPU_FIFO_WM)); outb(0x1, ICEREG1724(ice, MPU_FIFO_WM)); -#endif } } diff --git a/sound/sh/aica.c b/sound/sh/aica.c index d49417bf78c6..9ca113326143 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -663,7 +663,7 @@ static int __init aica_init(void) return err; pd = platform_device_register_simple(SND_AICA_DRIVER, -1, aica_memory_space, 2); - if (unlikely(IS_ERR(pd))) { + if (IS_ERR(pd)) { platform_driver_unregister(&snd_aica_driver); return PTR_ERR(pd); } diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index a3b51df2bea1..18f28ac4bfe8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -30,6 +30,7 @@ source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/davinci/Kconfig" +source "sound/soc/omap/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index e489dbdde458..782db2127108 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 76a5c7b05dfb..fb41826c4c4c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -150,7 +150,7 @@ static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - wm8753_write_reg_cache (codec, reg, value); + wm8753_write_reg_cache(codec, reg, value); if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -249,7 +249,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL); - if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0]) + if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0]) return 0; mode &= 0xfff3; @@ -342,7 +342,8 @@ static int wm8753_add_controls(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL)); + snd_soc_cnew(&wm8753_snd_controls[i], + codec, NULL)); if (err < 0) return err; } @@ -722,7 +723,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING - "WM8753 N value outwith recommended range! N = %d\n",Ndiv); + "wm8753: unsupported N = %d\n", Ndiv); pll_div->n = Ndiv; Nmod = target % source; @@ -1300,8 +1301,9 @@ static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) } #define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1507,9 +1509,9 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = socdev->codec; /* we only need to suspend if we are a valid card */ - if(!codec->card) + if (!codec->card) return 0; - + wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); return 0; } @@ -1523,7 +1525,7 @@ static int wm8753_resume(struct platform_device *pdev) u16 *cache = codec->reg_cache; /* we only need to resume if we are a valid card */ - if(!codec->card) + if (!codec->card) return 0; /* Sync reg_cache with the hardware */ @@ -1613,9 +1615,10 @@ static int wm8753_init(struct snd_soc_device *socdev) wm8753_add_widgets(codec); ret = snd_soc_register_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8753: failed to register card\n"); + printk(KERN_ERR "wm8753: failed to register card\n"); goto card_err; - } + } + return ret; card_err: @@ -1630,7 +1633,7 @@ pcm_err: around */ static struct snd_soc_device *wm8753_socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * WM8753 2 wire address is determined by GPIO5 @@ -1661,7 +1664,7 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL){ + if (!i2c) { kfree(codec); return -ENOMEM; } @@ -1749,7 +1752,7 @@ static int wm8753_probe(struct platform_device *pdev) wm8753_socdev = socdev; INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; @@ -1793,7 +1796,7 @@ static int wm8753_remove(struct platform_device *pdev) run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8753_i2c_driver); #endif kfree(codec->private_data); @@ -1808,7 +1811,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { .suspend = wm8753_suspend, .resume = wm8753_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); MODULE_DESCRIPTION("ASoC WM8753 driver"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index d2d79e182a45..76c1e2d33e7d 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -37,23 +37,23 @@ static int ac97_write(struct snd_soc_codec *codec, * WM9712 register cache */ static const u16 wm9712_reg[] = { - 0x6174, 0x8000, 0x8000, 0x8000, // 6 - 0x0f0f, 0xaaa0, 0xc008, 0x6808, // e - 0xe808, 0xaaa0, 0xad00, 0x8000, // 16 - 0xe808, 0x3000, 0x8000, 0x0000, // 1e - 0x0000, 0x0000, 0x0000, 0x000f, // 26 - 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e - 0x0000, 0xbb80, 0x0000, 0x0000, // 36 - 0x0000, 0x2000, 0x0000, 0x0000, // 3e - 0x0000, 0x0000, 0x0000, 0x0000, // 46 - 0x0000, 0x0000, 0xf83e, 0xffff, // 4e - 0x0000, 0x0000, 0x0000, 0xf83e, // 56 - 0x0008, 0x0000, 0x0000, 0x0000, // 5e - 0xb032, 0x3e00, 0x0000, 0x0000, // 66 - 0x0000, 0x0000, 0x0000, 0x0000, // 6e - 0x0000, 0x0000, 0x0000, 0x0006, // 76 - 0x0001, 0x0000, 0x574d, 0x4c12, // 7e - 0x0000, 0x0000 // virtual hp mixers + 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */ + 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */ + 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */ + 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */ + 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */ + 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ + 0x0000, 0x0000 /* virtual hp mixers */ }; /* virtual HP mixers regs */ @@ -94,7 +94,7 @@ static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = { SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), -SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1), +SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), @@ -165,7 +165,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL)); + snd_soc_cnew(&wm9712_snd_ac97_controls[i], + codec, NULL)); if (err < 0) return err; } @@ -363,7 +364,6 @@ static const char *audio_map[][3] = { {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"}, {"Left HP Mixer", NULL, "ALC Sidetone Mux"}, - //{"Right HP Mixer", NULL, "HP Mixer"}, /* Right HP mixer */ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, @@ -454,15 +454,13 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec) { int i; - for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); - } - /* set up audio path audio_mapnects */ - for(i = 0; audio_map[i][0] != NULL; i++) { + /* set up audio path connects */ + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + audio_map[i][1], audio_map[i][2]); snd_soc_dapm_new_widgets(codec); return 0; @@ -540,7 +538,8 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) } #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) struct snd_soc_codec_dai wm9712_dai[] = { { @@ -577,8 +576,6 @@ EXPORT_SYMBOL_GPL(wm9712_dai); static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) { - u16 reg; - switch (event) { case SNDRV_CTL_POWER_D0: /* full On */ case SNDRV_CTL_POWER_D1: /* partial On */ @@ -633,7 +630,7 @@ static int wm9712_soc_resume(struct platform_device *pdev) u16 *cache = codec->reg_cache; ret = wm9712_reset(codec, 1); - if (ret < 0){ + if (ret < 0) { printk(KERN_ERR "could not reset AC97 codec\n"); return ret; } @@ -642,9 +639,9 @@ static int wm9712_soc_resume(struct platform_device *pdev) if (ret == 0) { /* Sync reg_cache with the hardware after cold reset */ - for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) { + for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) { if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || - (i > 0x58 && i != 0x5c)) + (i > 0x58 && i != 0x5c)) continue; soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); } @@ -757,7 +754,6 @@ struct snd_soc_codec_device soc_codec_dev_wm9712 = { .suspend = wm9712_soc_suspend, .resume = wm9712_soc_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712); MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig new file mode 100644 index 000000000000..0230d83e8e5e --- /dev/null +++ b/sound/soc/omap/Kconfig @@ -0,0 +1,19 @@ +menu "SoC Audio for the Texas Instruments OMAP" + +config SND_OMAP_SOC + tristate "SoC Audio for the Texas Instruments OMAP chips" + depends on ARCH_OMAP && SND_SOC + +config SND_OMAP_SOC_MCBSP + tristate + select OMAP_MCBSP + +config SND_OMAP_SOC_N810 + tristate "SoC Audio support for Nokia N810" + depends on SND_OMAP_SOC && MACH_NOKIA_N810 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on Nokia N810. + +endmenu diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile new file mode 100644 index 000000000000..d8d8d58075e3 --- /dev/null +++ b/sound/soc/omap/Makefile @@ -0,0 +1,11 @@ +# OMAP Platform Support +snd-soc-omap-objs := omap-pcm.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o + +obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o +obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o + +# OMAP Machine Support +snd-soc-n810-objs := n810.o + +obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c new file mode 100644 index 000000000000..83b1eb4e40f3 --- /dev/null +++ b/sound/soc/omap/n810.c @@ -0,0 +1,336 @@ +/* + * n810.c -- SoC audio for Nokia N810 + * + * Copyright (C) 2008 Nokia Corporation + * + * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/hardware.h> +#include <asm/arch/gpio.h> +#include <asm/arch/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic3x.h" + +#define RX44_HEADSET_AMP_GPIO 10 +#define RX44_SPEAKER_AMP_GPIO 101 + +static struct clk *sys_clkout2; +static struct clk *sys_clkout2_src; +static struct clk *func96m_clk; + +static int n810_spk_func; +static int n810_jack_func; + +static void n810_ext_control(struct snd_soc_codec *codec) +{ + snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + + snd_soc_dapm_sync_endpoints(codec); +} + +static int n810_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->codec; + + n810_ext_control(codec); + return clk_enable(sys_clkout2); +} + +static void n810_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(sys_clkout2); +} + +static int n810_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = codec_dai->dai_ops.set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set cpu DAI configuration */ + err = cpu_dai->dai_ops.set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) + return err; + + /* Set the codec system clock for DAC and ADC */ + err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + SND_SOC_CLOCK_IN); + + return err; +} + +static struct snd_soc_ops n810_ops = { + .startup = n810_startup, + .hw_params = n810_hw_params, + .shutdown = n810_shutdown, +}; + +static int n810_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = n810_spk_func; + + return 0; +} + +static int n810_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (n810_spk_func == ucontrol->value.integer.value[0]) + return 0; + + n810_spk_func = ucontrol->value.integer.value[0]; + n810_ext_control(codec); + + return 1; +} + +static int n810_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = n810_jack_func; + + return 0; +} + +static int n810_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (n810_jack_func == ucontrol->value.integer.value[0]) + return 0; + + n810_jack_func = ucontrol->value.integer.value[0]; + n810_ext_control(codec); + + return 1; +} + +static int n810_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1); + else + omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0); + + return 0; +} + +static int n810_jack_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1); + else + omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0); + + return 0; +} + +static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event), + SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event), +}; + +static const char *audio_map[][3] = { + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + {"Ext Spk", NULL, "LLOUT"}, + {"Ext Spk", NULL, "RLOUT"}, +}; + +static const char *spk_function[] = {"Off", "On"}; +static const char *jack_function[] = {"Off", "Headphone"}; +static const struct soc_enum n810_enum[] = { + SOC_ENUM_SINGLE_EXT(2, spk_function), + SOC_ENUM_SINGLE_EXT(3, jack_function), +}; + +static const struct snd_kcontrol_new aic33_n810_controls[] = { + SOC_ENUM_EXT("Speaker Function", n810_enum[0], + n810_get_spk, n810_set_spk), + SOC_ENUM_EXT("Jack Function", n810_enum[1], + n810_get_jack, n810_set_jack), +}; + +static int n810_aic33_init(struct snd_soc_codec *codec) +{ + int i, err; + + /* Not connected */ + snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); + snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); + snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + + /* Add N810 specific controls */ + for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&aic33_n810_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + /* Add N810 specific widgets */ + for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]); + + /* Set up N810 specific audio path audio_map */ + for (i = 0; i < ARRAY_SIZE(audio_map); i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link n810_dai = { + .name = "TLV320AIC33", + .stream_name = "AIC33", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &aic3x_dai, + .init = n810_aic33_init, + .ops = &n810_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_n810 = { + .name = "N810", + .dai_link = &n810_dai, + .num_links = 1, +}; + +/* Audio private data */ +static struct aic3x_setup_data n810_aic33_setup = { + .i2c_address = 0x18, +}; + +/* Audio subsystem */ +static struct snd_soc_device n810_snd_devdata = { + .machine = &snd_soc_machine_n810, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &n810_aic33_setup, +}; + +static struct platform_device *n810_snd_device; + +static int __init n810_soc_init(void) +{ + int err; + struct device *dev; + + if (!machine_is_nokia_n810()) + return -ENODEV; + + n810_snd_device = platform_device_alloc("soc-audio", -1); + if (!n810_snd_device) + return -ENOMEM; + + platform_set_drvdata(n810_snd_device, &n810_snd_devdata); + n810_snd_devdata.dev = &n810_snd_device->dev; + *(unsigned int *)n810_dai.cpu_dai->private_data = 1; /* McBSP2 */ + err = platform_device_add(n810_snd_device); + if (err) + goto err1; + + dev = &n810_snd_device->dev; + + sys_clkout2_src = clk_get(dev, "sys_clkout2_src"); + if (IS_ERR(sys_clkout2_src)) { + dev_err(dev, "Could not get sys_clkout2_src clock\n"); + return -ENODEV; + } + sys_clkout2 = clk_get(dev, "sys_clkout2"); + if (IS_ERR(sys_clkout2)) { + dev_err(dev, "Could not get sys_clkout2\n"); + goto err1; + } + /* + * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use + * 96 MHz as its parent in order to get 12 MHz + */ + func96m_clk = clk_get(dev, "func_96m_ck"); + if (IS_ERR(func96m_clk)) { + dev_err(dev, "Could not get func 96M clock\n"); + goto err2; + } + clk_set_parent(sys_clkout2_src, func96m_clk); + clk_set_rate(sys_clkout2, 12000000); + + if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0) + BUG(); + if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0) + BUG(); + omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0); + omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0); + + return 0; +err2: + clk_put(sys_clkout2); + platform_device_del(n810_snd_device); +err1: + platform_device_put(n810_snd_device); + + return err; + +} + +static void __exit n810_soc_exit(void) +{ + platform_device_unregister(n810_snd_device); +} + +module_init(n810_soc_init); +module_exit(n810_soc_exit); + +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_DESCRIPTION("ALSA SoC Nokia N810"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c new file mode 100644 index 000000000000..40d87e6d0de8 --- /dev/null +++ b/sound/soc/omap/omap-mcbsp.c @@ -0,0 +1,414 @@ +/* + * omap-mcbsp.c -- OMAP ALSA SoC DAI driver using McBSP port + * + * Copyright (C) 2008 Nokia Corporation + * + * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/arch/control.h> +#include <asm/arch/dma.h> +#include <asm/arch/mcbsp.h> +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_KNOT) + +struct omap_mcbsp_data { + unsigned int bus_id; + struct omap_mcbsp_reg_cfg regs; + /* + * Flags indicating is the bus already activated and configured by + * another substream + */ + int active; + int configured; +}; + +#define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) + +static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; + +/* + * Stream DMA parameters. DMA request line and port address are set runtime + * since they are different between OMAP1 and later OMAPs + */ +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = { +{ + { .name = "I2S PCM Stereo out", }, + { .name = "I2S PCM Stereo in", }, +}, +}; + +#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) +static const int omap1_dma_reqs[][2] = { + { OMAP_DMA_MCBSP1_TX, OMAP_DMA_MCBSP1_RX }, + { OMAP_DMA_MCBSP2_TX, OMAP_DMA_MCBSP2_RX }, + { OMAP_DMA_MCBSP3_TX, OMAP_DMA_MCBSP3_RX }, +}; +static const unsigned long omap1_mcbsp_port[][2] = { + { OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, + OMAP1510_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, + { OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, + OMAP1510_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, + { OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DXR1, + OMAP1510_MCBSP3_BASE + OMAP_MCBSP_REG_DRR1 }, +}; +#else +static const int omap1_dma_reqs[][2] = {}; +static const unsigned long omap1_mcbsp_port[][2] = {}; +#endif +#if defined(CONFIG_ARCH_OMAP2420) +static const int omap2420_dma_reqs[][2] = { + { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, + { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, +}; +static const unsigned long omap2420_mcbsp_port[][2] = { + { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, + OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, + { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR1, + OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, +}; +#else +static const int omap2420_dma_reqs[][2] = {}; +static const unsigned long omap2420_mcbsp_port[][2] = {}; +#endif + +static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + int err = 0; + + if (!cpu_dai->active) + err = omap_mcbsp_request(mcbsp_data->bus_id); + + return err; +} + +static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + + if (!cpu_dai->active) { + omap_mcbsp_free(mcbsp_data->bus_id); + mcbsp_data->configured = 0; + } +} + +static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcbsp_data->active++) + omap_mcbsp_start(mcbsp_data->bus_id); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcbsp_data->active) + omap_mcbsp_stop(mcbsp_data->bus_id); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + unsigned long port; + + if (cpu_class_is_omap1()) { + dma = omap1_dma_reqs[bus_id][substream->stream]; + port = omap1_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap2420()) { + dma = omap2420_dma_reqs[bus_id][substream->stream]; + port = omap2420_mcbsp_port[bus_id][substream->stream]; + } else { + /* + * TODO: Add support for 2430 and 3430 + */ + return -ENODEV; + } + omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; + omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; + cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; + + if (mcbsp_data->configured) { + /* McBSP already configured by another stream */ + return 0; + } + + switch (params_channels(params)) { + case 2: + /* Set 1 word per (McBPSP) frame and use dual-phase frames */ + regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE; + regs->rcr1 |= RFRLEN1(1 - 1); + regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE; + regs->xcr1 |= XFRLEN1(1 - 1); + break; + default: + /* Unsupported number of channels */ + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + /* Set word lengths */ + regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); + regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); + regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); + regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); + /* Set FS period and length in terms of bit clock periods */ + regs->srgr2 |= FPER(16 * 2 - 1); + regs->srgr1 |= FWID(16 - 1); + break; + default: + /* Unsupported PCM format */ + return -EINVAL; + } + + omap_mcbsp_config(bus_id, &mcbsp_data->regs); + mcbsp_data->configured = 1; + + return 0; +} + +/* + * This must be called before _set_clkdiv and _set_sysclk since McBSP register + * cache is initialized here + */ +static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + + if (mcbsp_data->configured) + return 0; + + memset(regs, 0, sizeof(*regs)); + /* Generic McBSP register settings */ + regs->spcr2 |= XINTM(3) | FREE; + regs->spcr1 |= RINTM(3); + regs->rcr2 |= RFIG; + regs->xcr2 |= XFIG; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* 1-bit data delay */ + regs->rcr2 |= RDATDLY(1); + regs->xcr2 |= XDATDLY(1); + break; + default: + /* Unsupported data format */ + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* McBSP master. Set FS and bit clocks as outputs */ + regs->pcr0 |= FSXM | FSRM | + CLKXM | CLKRM; + /* Sample rate generator drives the FS */ + regs->srgr2 |= FSGM; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* McBSP slave */ + break; + default: + /* Unsupported master/slave configuration */ + return -EINVAL; + } + + /* Set bit clock (CLKX/CLKR) and FS polarities */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* + * Normal BCLK + FS. + * FS active low. TX data driven on falling edge of bit clock + * and RX data sampled on rising edge of bit clock. + */ + regs->pcr0 |= FSXP | FSRP | + CLKXP | CLKRP; + break; + case SND_SOC_DAIFMT_NB_IF: + regs->pcr0 |= CLKXP | CLKRP; + break; + case SND_SOC_DAIFMT_IB_NF: + regs->pcr0 |= FSXP | FSRP; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + + if (div_id != OMAP_MCBSP_CLKGDV) + return -ENODEV; + + regs->srgr1 |= CLKGDV(div - 1); + + return 0; +} + +static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, + int clk_id) +{ + int sel_bit; + u16 reg; + + if (cpu_class_is_omap1()) { + /* OMAP1's can use only external source clock */ + if (unlikely(clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)) + return -EINVAL; + else + return 0; + } + + switch (mcbsp_data->bus_id) { + case 0: + reg = OMAP2_CONTROL_DEVCONF0; + sel_bit = 2; + break; + case 1: + reg = OMAP2_CONTROL_DEVCONF0; + sel_bit = 6; + break; + /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */ + default: + return -EINVAL; + } + + if (cpu_class_is_omap2()) { + if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) { + omap_ctrl_writel(omap_ctrl_readl(reg) & + ~(1 << sel_bit), reg); + } else { + omap_ctrl_writel(omap_ctrl_readl(reg) | + (1 << sel_bit), reg); + } + } + + return 0; +} + +static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, + int dir) +{ + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + int err = 0; + + switch (clk_id) { + case OMAP_MCBSP_SYSCLK_CLK: + regs->srgr2 |= CLKSM; + break; + case OMAP_MCBSP_SYSCLK_CLKS_FCLK: + case OMAP_MCBSP_SYSCLK_CLKS_EXT: + err = omap_mcbsp_dai_set_clks_src(mcbsp_data, clk_id); + break; + + case OMAP_MCBSP_SYSCLK_CLKX_EXT: + regs->srgr2 |= CLKSM; + case OMAP_MCBSP_SYSCLK_CLKR_EXT: + regs->pcr0 |= SCLKME; + break; + default: + err = -ENODEV; + } + + return err; +} + +struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = { +{ + .name = "omap-mcbsp-dai", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = OMAP_MCBSP_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = OMAP_MCBSP_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .startup = omap_mcbsp_dai_startup, + .shutdown = omap_mcbsp_dai_shutdown, + .trigger = omap_mcbsp_dai_trigger, + .hw_params = omap_mcbsp_dai_hw_params, + }, + .dai_ops = { + .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, + }, + .private_data = &mcbsp_data[0].bus_id, +}, +}; +EXPORT_SYMBOL_GPL(omap_mcbsp_dai); + +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_DESCRIPTION("OMAP I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h new file mode 100644 index 000000000000..9965fd4b0427 --- /dev/null +++ b/sound/soc/omap/omap-mcbsp.h @@ -0,0 +1,49 @@ +/* + * omap-mcbsp.h + * + * Copyright (C) 2008 Nokia Corporation + * + * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_I2S_H__ +#define __OMAP_I2S_H__ + +/* Source clocks for McBSP sample rate generator */ +enum omap_mcbsp_clksrg_clk { + OMAP_MCBSP_SYSCLK_CLKS_FCLK, /* Internal FCLK */ + OMAP_MCBSP_SYSCLK_CLKS_EXT, /* External CLKS pin */ + OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */ + OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */ + OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */ +}; + +/* McBSP dividers */ +enum omap_mcbsp_div { + OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ +}; + +/* + * REVISIT: Preparation for the ASoC v2. Let the number of available links to + * be same than number of McBSP ports found in OMAP(s) we are compiling for. + */ +#define NUM_LINKS 1 + +extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS]; + +#endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c new file mode 100644 index 000000000000..62370202c649 --- /dev/null +++ b/sound/soc/omap/omap-pcm.c @@ -0,0 +1,357 @@ +/* + * omap-pcm.c -- ALSA PCM interface for the OMAP SoC + * + * Copyright (C) 2008 Nokia Corporation + * + * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/arch/dma.h> +#include "omap-pcm.h" + +static const struct snd_pcm_hardware omap_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 64 * 1024, + .periods_min = 2, + .periods_max = 255, + .buffer_bytes_max = 128 * 1024, +}; + +struct omap_runtime_data { + spinlock_t lock; + struct omap_pcm_dma_data *dma_data; + int dma_ch; + int period_index; +}; + +static void omap_pcm_dma_irq(int ch, u16 stat, void *data) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd = runtime->private_data; + unsigned long flags; + + if (cpu_is_omap1510()) { + /* + * OMAP1510 doesn't support DMA chaining so have to restart + * the transfer after all periods are transferred + */ + spin_lock_irqsave(&prtd->lock, flags); + if (prtd->period_index >= 0) { + if (++prtd->period_index == runtime->periods) { + prtd->period_index = 0; + omap_start_dma(prtd->dma_ch); + } + } + spin_unlock_irqrestore(&prtd->lock, flags); + } + + snd_pcm_period_elapsed(substream); +} + +/* this may get called several times by oss emulation */ +static int omap_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct omap_runtime_data *prtd = runtime->private_data; + struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; + int err = 0; + + if (!dma_data) + return -ENODEV; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + if (prtd->dma_data) + return 0; + prtd->dma_data = dma_data; + err = omap_request_dma(dma_data->dma_req, dma_data->name, + omap_pcm_dma_irq, substream, &prtd->dma_ch); + if (!cpu_is_omap1510()) { + /* + * Link channel with itself so DMA doesn't need any + * reprogramming while looping the buffer + */ + omap_dma_link_lch(prtd->dma_ch, prtd->dma_ch); + } + + return err; +} + +static int omap_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd = runtime->private_data; + + if (prtd->dma_data == NULL) + return 0; + + if (!cpu_is_omap1510()) + omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); + omap_free_dma(prtd->dma_ch); + prtd->dma_data = NULL; + + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int omap_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd = runtime->private_data; + struct omap_pcm_dma_data *dma_data = prtd->dma_data; + struct omap_dma_channel_params dma_params; + + memset(&dma_params, 0, sizeof(dma_params)); + /* + * Note: Regardless of interface data formats supported by OMAP McBSP + * or EAC blocks, internal representation is always fixed 16-bit/sample + */ + dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.trigger = dma_data->dma_req; + dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_params.src_amode = OMAP_DMA_AMODE_POST_INC; + dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT; + dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; + dma_params.src_start = runtime->dma_addr; + dma_params.dst_start = dma_data->port_addr; + } else { + dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; + dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; + dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; + dma_params.src_start = dma_data->port_addr; + dma_params.dst_start = runtime->dma_addr; + } + /* + * Set DMA transfer frame size equal to ALSA period size and frame + * count as no. of ALSA periods. Then with DMA frame interrupt enabled, + * we can transfer the whole ALSA buffer with single DMA transfer but + * still can get an interrupt at each period bounary + */ + dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + dma_params.frame_count = runtime->periods; + omap_set_dma_params(prtd->dma_ch, &dma_params); + + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + + return 0; +} + +static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd = runtime->private_data; + int ret = 0; + + spin_lock_irq(&prtd->lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->period_index = 0; + omap_start_dma(prtd->dma_ch); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->period_index = -1; + omap_stop_dma(prtd->dma_ch); + break; + default: + ret = -EINVAL; + } + spin_unlock_irq(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd = runtime->private_data; + dma_addr_t ptr; + snd_pcm_uframes_t offset; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ptr = omap_get_dma_src_pos(prtd->dma_ch); + else + ptr = omap_get_dma_dst_pos(prtd->dma_ch); + + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int omap_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct omap_runtime_data *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); + + /* Ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + prtd = kzalloc(sizeof(prtd), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + spin_lock_init(&prtd->lock); + runtime->private_data = prtd; + +out: + return ret; +} + +static int omap_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + kfree(runtime->private_data); + return 0; +} + +static int omap_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops omap_pcm_ops = { + .open = omap_pcm_open, + .close = omap_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = omap_pcm_hw_params, + .hw_free = omap_pcm_hw_free, + .prepare = omap_pcm_prepare, + .trigger = omap_pcm_trigger, + .pointer = omap_pcm_pointer, + .mmap = omap_pcm_mmap, +}; + +static u64 omap_pcm_dmamask = DMA_BIT_MASK(32); + +static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = omap_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &omap_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (dai->playback.channels_min) { + ret = omap_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = omap_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +struct snd_soc_platform omap_soc_platform = { + .name = "omap-pcm-audio", + .pcm_ops = &omap_pcm_ops, + .pcm_new = omap_pcm_new, + .pcm_free = omap_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(omap_soc_platform); + +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); +MODULE_DESCRIPTION("OMAP PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h new file mode 100644 index 000000000000..e4369bdfd77d --- /dev/null +++ b/sound/soc/omap/omap-pcm.h @@ -0,0 +1,35 @@ +/* + * omap-pcm.h + * + * Copyright (C) 2008 Nokia Corporation + * + * Contact: Jarkko Nikula <jarkko.nikula@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_PCM_H__ +#define __OMAP_PCM_H__ + +struct omap_pcm_dma_data { + char *name; /* stream identifier */ + int dma_req; /* DMA request line */ + unsigned long port_addr; /* transmit/receive register */ +}; + +extern struct snd_soc_platform omap_soc_platform; + +#endif diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 9ed8f2e8da10..4eab2c19c454 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -1,10 +1,10 @@ /* * SoC audio for ln2440sbc - * + * * Copyright 2007 KonekTel, a.s. * Author: Ivan Kuten * ivan.kuten@promwad.com - * + * * Heavily based on smdk2443_wm9710.c * Copyright 2007 Wolfson Microelectronics PLC. * Author: Graeme Gregory diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 962cc20b1af5..0e9d1c5f2484 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -33,7 +33,7 @@ #include <asm/arch/regs-gpio.h> #include <asm/hardware.h> #include <asm/arch/audio.h> -#include <asm/io.h> +#include <linux/io.h> #include <asm/arch/spi-gpio.h> #include <asm/plat-s3c24xx/regs-iis.h> @@ -122,7 +122,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, /* set MCLK division for sample rate */ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS ); + S3C2410_IISMOD_32FS); if (ret < 0) return ret; @@ -133,7 +133,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, /* set prescaler division for sample rate */ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4,4)); + S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; @@ -222,7 +222,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario = 0; +static int neo1973_scenario; static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -233,7 +233,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { - switch(neo1973_scenario) { + switch (neo1973_scenario) { case NEO_AUDIO_OFF: snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); @@ -334,7 +334,7 @@ static void lm4857_write_regs(void) static int lm4857_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int reg=kcontrol->private_value & 0xFF; + int reg = kcontrol->private_value & 0xFF; int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; @@ -349,11 +349,11 @@ static int lm4857_set_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; - if (((lm4857_regs[reg] >> shift ) & mask) == + if (((lm4857_regs[reg] >> shift) & mask) == ucontrol->value.integer.value[0]) return 0; - lm4857_regs[reg] &= ~ (mask << shift); + lm4857_regs[reg] &= ~(mask << shift); lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; lm4857_write_regs(); return 1; @@ -398,7 +398,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { /* example machine audio_mapnections */ -static const char* audio_map[][3] = { +static const char *audio_map[][3] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -450,7 +450,7 @@ static const char *neo_scenarios[] = { }; static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios),neo_scenarios), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { @@ -521,8 +521,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = -{ .name = "Bluetooth", +static struct snd_soc_cpu_dai bt_dai = { + .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, .playback = { @@ -616,6 +616,35 @@ static int lm4857_i2c_attach(struct i2c_adapter *adap) return i2c_probe(adap, &addr_data, lm4857_amp_probe); } +static u8 lm4857_state; + +static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) +{ + dev_dbg(&dev->dev, "lm4857_suspend\n"); + lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); + } + return 0; +} + +static int lm4857_resume(struct i2c_client *dev) +{ + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); + lm4857_write_regs(); + } + return 0; +} + +static void lm4857_shutdown(struct i2c_client *dev) +{ + dev_dbg(&dev->dev, "lm4857_shutdown\n"); + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); +} + /* corgi i2c codec control layer */ static struct i2c_driver lm4857_i2c_driver = { .driver = { @@ -623,6 +652,9 @@ static struct i2c_driver lm4857_i2c_driver = { .owner = THIS_MODULE, }, .id = I2C_DRIVERID_LM4857, + .suspend = lm4857_suspend, + .resume = lm4857_resume, + .shutdown = lm4857_shutdown, .attach_adapter = lm4857_i2c_attach, .detach_client = lm4857_i2c_detach, .command = NULL, @@ -667,6 +699,6 @@ module_init(neo1973_init); module_exit(neo1973_exit); /* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org"); MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 1c1ddbf7f3c0..e81d9a6c83da 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/io.h> #include <linux/wait.h> #include <linux/delay.h> #include <linux/clk.h> @@ -30,7 +31,6 @@ #include <sound/soc.h> #include <asm/hardware.h> -#include <asm/io.h> #include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> @@ -47,7 +47,7 @@ struct s3c24xx_ac97_info { }; static struct s3c24xx_ac97_info s3c24xx_ac97; -DECLARE_COMPLETION(ac97_completion); +static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; static DECLARE_MUTEX(ac97_mutex); @@ -290,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) u32 ac_glbctrl; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch(cmd) { + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -333,7 +333,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, u32 ac_glbctrl; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - switch(cmd) { + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -391,7 +391,6 @@ struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { .trigger = s3c2443_ac97_mic_trigger,}, }, }; - EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4ebcd6a8bf28..1ed6afd45459 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -224,6 +224,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, iismod |= S3C2410_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2410_IISMOD_SLAVE; break; default: return -EINVAL; @@ -234,6 +235,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, iismod |= S3C2410_IISMOD_MSB; break; case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2410_IISMOD_MSB; break; default: return -EINVAL; diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 49580fb481d5..7806ae614617 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -20,6 +20,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/io.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dma-mapping.h> @@ -30,7 +31,6 @@ #include <sound/soc.h> #include <asm/dma.h> -#include <asm/io.h> #include <asm/hardware.h> #include <asm/arch/dma.h> #include <asm/arch/audio.h> @@ -93,7 +93,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) while (prtd->dma_loaded < prtd->dma_limit) { unsigned long len = prtd->dma_period; - DBG("dma_loaded: %d\n",prtd->dma_loaded); + DBG("dma_loaded: %d\n", prtd->dma_loaded); if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; @@ -101,7 +101,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) __func__, len); } - ret = s3c2410_dma_enqueue(prtd->params->channel, + ret = s3c2410_dma_enqueue(prtd->params->channel, substream, pos, len); if (ret == 0) { @@ -129,7 +129,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, return; prtd = substream->runtime->private_data; - + if (substream) snd_pcm_period_elapsed(substream); @@ -150,7 +150,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); - int ret=0; + int ret = 0; DBG("Entered %s\n", __func__); @@ -171,7 +171,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, ret = s3c2410_dma_request(prtd->params->channel, prtd->params->client, NULL); - if (ret) { + if (ret < 0) { DBG(KERN_ERR "failed to get dma channel\n"); return ret; } @@ -223,7 +223,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!prtd->params) - return 0; + return 0; /* channel needs configuring for mem=>device, increment memory addr, * sync to pclk, half-word transfers to the IIS-FIFO. */ @@ -293,8 +293,8 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static snd_pcm_uframes_t - s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t +s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -313,7 +313,7 @@ static snd_pcm_uframes_t spin_unlock(&prtd->lock); - DBG("Pointer %x %x\n",src,dst); + DBG("Pointer %x %x\n", src, dst); /* we seem to be getting the odd error from the pcm library due * to out-of-bounds pointers. this is maybe due to the dma engine @@ -355,11 +355,11 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); - if (prtd) - kfree(prtd); - else + if (!prtd) DBG("s3c24xx_pcm_close called with prtd == NULL\n"); + kfree(prtd); + return 0; } @@ -371,9 +371,9 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, DBG("Entered %s\n", __func__); return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); } static struct snd_pcm_ops s3c24xx_pcm_ops = { @@ -432,7 +432,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -467,7 +467,6 @@ struct snd_soc_platform s3c24xx_soc_platform = { .pcm_new = s3c24xx_pcm_new, .pcm_free = s3c24xx_pcm_free_dma_buffers, }; - EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); |