diff options
Diffstat (limited to 'sound')
48 files changed, 1121 insertions, 292 deletions
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 7793d2a511ce..0afd1a8226fb 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -238,6 +238,8 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); @@ -254,10 +256,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } @@ -268,6 +270,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); @@ -285,9 +289,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 28b3c7f7cfe6..f2f41c854221 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -165,7 +165,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = { .buffer_bytes_max = 64 * 4096, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; @@ -502,7 +502,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n", - dac->regs, dac->dma.chan->dev->device.bus_id); + dac->regs, dev_name(&dac->dma.chan->dev->device)); return retval; diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index dd72e00e5ae1..0c0f8771656a 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1,5 +1,5 @@ /* - * Driver for the Atmel AC97C controller + * Driver for Atmel AC97C * * Copyright (C) 2005-2009 Atmel Corporation * @@ -10,6 +10,7 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/bitmap.h> +#include <linux/device.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> #include <linux/init.h> @@ -65,6 +66,7 @@ struct atmel_ac97c { /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; + int irq; int opened; int reset_pin; }; @@ -150,10 +152,10 @@ static struct snd_pcm_hardware atmel_ac97c_hw = { .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = 64 * 4096, + .buffer_bytes_max = 2 * 2 * 64 * 2048, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; @@ -297,9 +299,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long word = 0; + unsigned long word = ac97c_readl(chip, OCA); int retval; + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + /* assign channels to AC97C channel A */ switch (runtime->channels) { case 1: @@ -312,7 +316,6 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) default: /* TODO: support more than two channels */ return -EINVAL; - break; } ac97c_writel(chip, OCA, word); @@ -324,13 +327,25 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ - default: word &= ~(AC97C_CMR_CEM_LITTLE); break; + default: + word = ac97c_readl(chip, OCA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, OCA, word); + return -EINVAL; } + /* Enable underrun interrupt on channel A */ + word |= AC97C_CSR_UNRUN; + ac97c_writel(chip, CAMR, word); + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + /* set variable rate if needed */ if (runtime->rate != 48000) { word = ac97c_readl(chip, MR); @@ -359,9 +374,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long word = 0; + unsigned long word = ac97c_readl(chip, ICA); int retval; + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + /* assign channels to AC97C channel A */ switch (runtime->channels) { case 1: @@ -374,7 +391,6 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) default: /* TODO: support more than two channels */ return -EINVAL; - break; } ac97c_writel(chip, ICA, word); @@ -386,13 +402,25 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ - default: word &= ~(AC97C_CMR_CEM_LITTLE); break; + default: + word = ac97c_readl(chip, ICA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, ICA, word); + return -EINVAL; } + /* Enable overrun interrupt on channel A */ + word |= AC97C_CSR_OVRUN; + ac97c_writel(chip, CAMR, word); + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + /* set variable rate if needed */ if (runtime->rate != 48000) { word = ac97c_readl(chip, MR); @@ -543,6 +571,43 @@ static struct snd_pcm_ops atmel_ac97_capture_ops = { .pointer = atmel_ac97c_capture_pointer, }; +static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) +{ + struct atmel_ac97c *chip = (struct atmel_ac97c *)dev; + irqreturn_t retval = IRQ_NONE; + u32 sr = ac97c_readl(chip, SR); + u32 casr = ac97c_readl(chip, CASR); + u32 cosr = ac97c_readl(chip, COSR); + + if (sr & AC97C_SR_CAEVT) { + dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + casr & AC97C_CSR_OVRUN ? " OVRUN" : "", + casr & AC97C_CSR_RXRDY ? " RXRDY" : "", + casr & AC97C_CSR_UNRUN ? " UNRUN" : "", + casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + casr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !casr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (sr & AC97C_SR_COEVT) { + dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", + cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", + cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + cosr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !cosr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (retval == IRQ_NONE) { + dev_err(&chip->pdev->dev, "spurious interrupt sr 0x%08x " + "casr 0x%08x cosr 0x%08x\n", sr, casr, cosr); + } + + return retval; +} + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; @@ -665,17 +730,17 @@ static bool filter(struct dma_chan *chan, void *slave) static void atmel_ac97c_reset(struct atmel_ac97c *chip) { - ac97c_writel(chip, MR, AC97C_MR_WRST); + ac97c_writel(chip, MR, 0); + ac97c_writel(chip, MR, AC97C_MR_ENA); + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); if (gpio_is_valid(chip->reset_pin)) { gpio_set_value(chip->reset_pin, 0); /* AC97 v2.2 specifications says minimum 1 us. */ - udelay(10); + udelay(2); gpio_set_value(chip->reset_pin, 1); } - - udelay(1); - ac97c_writel(chip, MR, AC97C_MR_ENA); } static int __devinit atmel_ac97c_probe(struct platform_device *pdev) @@ -690,6 +755,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) .read = atmel_ac97c_read, }; int retval; + int irq; regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!regs) { @@ -703,6 +769,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get irq\n"); + return -ENXIO; + } + pclk = clk_get(&pdev->dev, "pclk"); if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); @@ -719,6 +791,13 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip = get_chip(card); + retval = request_irq(irq, atmel_ac97c_interrupt, 0, "AC97C", chip); + if (retval) { + dev_dbg(&pdev->dev, "unable to request irq %d\n", irq); + goto err_request_irq; + } + chip->irq = irq; + spin_lock_init(&chip->lock); strcpy(card->driver, "Atmel AC97C"); @@ -747,14 +826,18 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) snd_card_set_dev(card, &pdev->dev); + atmel_ac97c_reset(chip); + + /* Enable overrun interrupt from codec channel */ + ac97c_writel(chip, COMR, AC97C_CSR_OVRUN); + ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT); + retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); if (retval) { dev_dbg(&pdev->dev, "could not register on ac97 bus\n"); goto err_ac97_bus; } - atmel_ac97c_reset(chip); - retval = atmel_ac97c_mixer_new(chip); if (retval) { dev_dbg(&pdev->dev, "could not register ac97 mixer\n"); @@ -773,7 +856,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip->dma.rx_chan = dma_request_channel(mask, filter, dws); dev_info(&chip->pdev->dev, "using %s for DMA RX\n", - chip->dma.rx_chan->dev->device.bus_id); + dev_name(&chip->dma.rx_chan->dev->device)); set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); } @@ -789,7 +872,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip->dma.tx_chan = dma_request_channel(mask, filter, dws); dev_info(&chip->pdev->dev, "using %s for DMA TX\n", - chip->dma.tx_chan->dev->device.bus_id); + dev_name(&chip->dma.tx_chan->dev->device)); set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); } @@ -809,7 +892,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) retval = snd_card_register(card); if (retval) { dev_dbg(&pdev->dev, "could not register sound card\n"); - goto err_ac97_bus; + goto err_dma; } platform_set_drvdata(pdev, card); @@ -836,6 +919,8 @@ err_ac97_bus: iounmap(chip->regs); err_ioremap: + free_irq(irq, chip); +err_request_irq: snd_card_free(card); err_snd_card_new: clk_disable(pclk); @@ -884,9 +969,14 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) if (gpio_is_valid(chip->reset_pin)) gpio_free(chip->reset_pin); + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); + ac97c_writel(chip, MR, 0); + clk_disable(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); + free_irq(chip->irq, chip); if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) dma_release_channel(chip->dma.rx_chan); diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h index c17bd5825980..ecbba5021c80 100644 --- a/sound/atmel/ac97c.h +++ b/sound/atmel/ac97c.h @@ -1,5 +1,5 @@ /* - * Register definitions for the Atmel AC97C controller + * Register definitions for Atmel AC97C * * Copyright (C) 2005-2009 Atmel Corporation * @@ -17,10 +17,6 @@ #define AC97C_CATHR 0x24 #define AC97C_CASR 0x28 #define AC97C_CAMR 0x2c -#define AC97C_CBRHR 0x30 -#define AC97C_CBTHR 0x34 -#define AC97C_CBSR 0x38 -#define AC97C_CBMR 0x3c #define AC97C_CORHR 0x40 #define AC97C_COTHR 0x44 #define AC97C_COSR 0x48 @@ -46,8 +42,10 @@ #define AC97C_MR_VRA (1 << 2) #define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_TXEMPTY (1 << 1) #define AC97C_CSR_UNRUN (1 << 2) #define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_OVRUN (1 << 5) #define AC97C_CSR_ENDTX (1 << 10) #define AC97C_CSR_ENDRX (1 << 14) @@ -61,11 +59,15 @@ #define AC97C_CMR_DMAEN (1 << 22) #define AC97C_SR_CAEVT (1 << 3) +#define AC97C_SR_COEVT (1 << 2) +#define AC97C_SR_WKUP (1 << 1) +#define AC97C_SR_SOF (1 << 0) +#define AC97C_CH_MASK(slot) \ + (0x7 << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CH_ASSIGN(slot, channel) \ (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CHANNEL_NONE 0x0 #define AC97C_CHANNEL_A 0x1 -#define AC97C_CHANNEL_B 0x2 #endif /* __SOUND_ATMEL_AC97C_H */ diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index e570649184e2..5dcd8a526970 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -703,19 +703,27 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, if (left || right) { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } else { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index ef95279da7a3..0481a55334b9 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -481,6 +481,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1, OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1), OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1, db_scale_5bit_12db_max), +OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0), }; static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = { diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index d1f242bd0ac5..8f5098f92c37 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -909,8 +909,8 @@ snd_ad1889_create(struct snd_card *card, return err; /* check PCI availability (32bit DMA) */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR PFX "error setting 32-bit DMA mask.\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 4edf270a7809..c551006e2920 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2186,8 +2186,8 @@ static int __devinit snd_ali_create(struct snd_card *card, if (err < 0) return err; /* check, if we can restrict PCI DMA transfers to 31 bits */ - if (pci_set_dma_mask(pci, DMA_31BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_31BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(31)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(31)) < 0) { snd_printk(KERN_ERR "architecture does not support " "31bit PCI busmaster DMA\n"); pci_disable_device(pci); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 009b4c8225a5..3aa35af7ca91 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -689,8 +689,8 @@ static int __devinit snd_als300_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { printk(KERN_ERR "error setting 28bit DMA mask\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 542a0c65a92c..3dbacde1a5af 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -872,8 +872,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, return err; } /* 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) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 9ec122383eef..7b72c88e449d 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -151,8 +151,8 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) // check PCI availability (DMA). if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR "error to set DMA mask\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 8eea29fc42fe..4d34bb0d99d3 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -279,8 +279,8 @@ static int __devinit snd_aw2_create(struct snd_card *card, pci_set_master(pci); /* check PCI availability (32bit DMA) */ - if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || - (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + if ((pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0)) { printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index e9e9b5821d41..f290bc56178f 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2125,8 +2125,8 @@ snd_azf3328_create(struct snd_card *card, chip->irq = -1; /* 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) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support " "24bit PCI busmaster DMA\n" ); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index df757575798a..bfac30f7929f 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1589,8 +1589,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, err = pci_enable_device(pci); if (err < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR "error to set 32bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index c89ed1f5bc2b..05f56e04849b 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -285,8 +285,8 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_WARNING "unable to get 32bit dma\n"); err = -ENXIO; goto pcifail; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 31542adc6b7e..1970f0e70f37 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -897,8 +897,8 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "error to set 28bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index dd63b132fb8e..fbd2ac09aa34 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1608,8 +1608,8 @@ static int __devinit snd_es1938_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* 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) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index dc97e8116141..a11f453a6b6d 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2539,8 +2539,8 @@ static int __devinit snd_es1968_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5bb48ee8b6c6..38ad3f7b040f 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3256,7 +3256,7 @@ static const char *ad1884_slave_vols[] = { "Mic Playback Volume", "CD Playback Volume", "Internal Mic Playback Volume", - "Docking Mic Playback Volume" + "Docking Mic Playback Volume", /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82097790f6f3..f35e58a2d921 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { {} }; +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + static hda_nid_t alc1200_slave_dig_outs[] = { ALC883_DIGOUT_NID, 0, }; @@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), .channel_mode = alc883_3ST_6ch_intel_modes, .need_dac_fix = 1, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b5e108aa8f63..61996a2f45df 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4413,6 +4413,24 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); stac92xx_report_jack(codec, event->nid); + + switch (codec->subsystem_id) { + case 0x103c308f: + if (event->nid == 0xb) { + int pin = AC_PINCTL_IN_EN; + + if (get_pin_presence(codec, 0xa) + && get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + if (!get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + + /* toggle VREF state based on mic + hp pin + * status + */ + stac92xx_auto_set_pinctl(codec, 0x0a, pin); + } + } break; case STAC_VREF_EVENT: data = snd_hda_codec_read(codec, codec->afg, 0, @@ -4895,6 +4913,7 @@ again: switch (codec->vendor_id) { case 0x111d7604: case 0x111d7605: + case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; @@ -5707,6 +5726,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 3dd63f1cda53..0d0cdbdb4486 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2533,8 +2533,8 @@ static int __devinit snd_ice1712_create(struct snd_card *card, if (err < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 70141548f251..75283fbb4b3f 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2530,8 +2530,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, return -EIO; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index c1eb84a14c42..82bc5b9e7629 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1291,7 +1291,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 80e064a3efff..833e9c7b27c7 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1449,7 +1449,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { snd_printk(KERN_ERR "architecture does not support " "32bit PCI busmaster DMA\n"); pci_disable_device(pci); diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index baf6d8e3dabc..1a5ff0611072 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1300,7 +1300,7 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { printk(KERN_ERR "sis7019: architecture does not support " "30-bit PCI busmaster DMA"); goto error_out_enabled; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index d989215f3556..7dc60ad4772e 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1264,8 +1264,8 @@ static int __devinit snd_sonicvibes_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* 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) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index a9da9c184660..6d943f6f6b70 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3559,8 +3559,8 @@ int __devinit snd_trident_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 30 bits */ - if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_30BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { snd_printk(KERN_ERR "architecture does not support 30bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 5a929069dce9..a2b69b8cff43 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -51,7 +51,7 @@ static struct platform_device *device; /* */ -static int __init snd_pmac_probe(struct platform_device *devptr) +static int __devinit snd_pmac_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_pmac *chip; diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 8cfed1a5dcbe..cf0dfb7ca221 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -413,7 +413,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -424,7 +424,7 @@ int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 1318c4f627b7..62fbb8459569 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -245,7 +245,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -256,7 +256,7 @@ int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 1f63d387a2f4..dd3380202766 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -659,7 +659,8 @@ static int ak4535_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); + if (codec->control_data) + i2c_unregister_device(codec->control_data); i2c_del_driver(&ak4535_i2c_driver); #endif kfree(codec->private_data); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 97738e2ece04..921b205de28a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -122,6 +122,9 @@ struct twl4030_priv { unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; }; /* @@ -1217,6 +1220,52 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int twl4030_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + /* If we already have a playback or capture going then constrain + * this substream to match it. + */ + if (twl4030->master_substream) { + struct snd_pcm_runtime *master_runtime; + master_runtime = twl4030->master_substream->runtime; + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + twl4030->slave_substream = substream; + } else + twl4030->master_substream = substream; + + return 0; +} + +static void twl4030_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + if (twl4030->master_substream == substream) + twl4030->master_substream = twl4030->slave_substream; + + twl4030->slave_substream = NULL; +} + static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1224,8 +1273,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; + if (substream == twl4030->slave_substream) + /* Ignoring hw_params for slave substream */ + return 0; + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1259,6 +1313,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); @@ -1384,6 +1441,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) static struct snd_soc_dai_ops twl4030_dai_ops = { + .startup = twl4030_startup, + .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, @@ -1395,7 +1454,7 @@ struct snd_soc_dai twl4030_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = TWL4030_RATES, + .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 33dbb144dad1..cb63765db1df 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -109,6 +109,7 @@ #define TWL4030_APLL_RATE_32000 0x80 #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3265817c5c26..6e23a81dba78 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec) return -EIO; } +#ifdef CONFIG_PM +static int wm9705_soc_suspend(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + + return 0; +} + +static int wm9705_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9705_reset(codec); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + + return 0; +} +#else +#define wm9705_soc_suspend NULL +#define wm9705_soc_resume NULL +#endif + static int wm9705_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, + .suspend = wm9705_soc_suspend, + .resume = wm9705_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb8570cd7b..b1a3a278819f 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -300,7 +300,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, return -ENOMEM; } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { @@ -418,7 +418,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return -EBUSY; } - dma_private = dma_alloc_coherent(substream->pcm->dev, + dma_private = dma_alloc_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); if (!dma_private) { dev_err(substream->pcm->card->dev, @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; @@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + /* + * When capture is started, the SSI immediately starts to fill its FIFO. + * This means that the DMA controller is not started until the FIFO is + * full. However, ALSA calls this function before that happens, when + * MR.DAR is still zero. In this case, just return zero to indicate + * that nothing has been received yet. + */ + if (!position) + return 0; + + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "dma pointer is out of range, halting stream\n"); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* @@ -761,13 +778,13 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) free_irq(dma_private->irq, dma_private); if (dma_private->ld_buf_phys) { - dma_unmap_single(substream->pcm->dev, + dma_unmap_single(substream->pcm->card->dev, dma_private->ld_buf_phys, sizeof(dma_private->link), DMA_TO_DEVICE); } /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); substream->runtime->private_data = NULL; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 169bca295b78..3711d8454d96 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -60,6 +60,13 @@ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) #endif +/* SIER bitflag of interrupts to enable */ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) + /** * fsl_ssi_private: per-SSI private data * @@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + sisr = in_be32(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* 4. Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, - CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + out_be32(&ssi->sier, SIER_FLAGS); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -466,28 +468,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - } else { - long timeout = jiffies + 10; - + else setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - - /* Wait until the SSI has filled its FIFO. Without this - * delay, ALSA complains about overruns. When the FIFO - * is full, the DMA controller initiates its first - * transfer. Until then, however, the DMA's DAR - * register is zero, which translates to an - * out-of-bounds pointer. This makes ALSA think an - * overrun has occurred. - */ - while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && - (jiffies < timeout)); - if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) - return -EIO; - } break; case SNDRV_PCM_TRIGGER_STOP: @@ -606,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .ops = &fsl_ssi_dai_ops, }; +/* Show the statistics of a flag only if its interrupt is enabled. The + * compiler will optimze this code to a no-op if the interrupt is not + * enabled. + */ +#define SIER_SHOW(flag, name) \ + do { \ + if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ + length += sprintf(buf + length, #name "=%u\n", \ + ssi_private->stats.name); \ + } while (0) + + /** * fsl_sysfs_ssi_show: display SSI statistics * - * Display the statistics for the current SSI device. + * Display the statistics for the current SSI device. To avoid confusion, + * we only show those counts that are enabled. */ static ssize_t fsl_sysfs_ssi_show(struct device *dev, struct device_attribute *attr, char *buf) { struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length; - - length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); - length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); - length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); - length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); - length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); - length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); - length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); - length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); - length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); - length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); - length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); - length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); - length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); - length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); - length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); - length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); - length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); - length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); - length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); - length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); - length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length = 0; + + SIER_SHOW(RFRC_EN, rfrc); + SIER_SHOW(TFRC_EN, tfrc); + SIER_SHOW(CMDAU_EN, cmdau); + SIER_SHOW(CMDDU_EN, cmddu); + SIER_SHOW(RXT_EN, rxt); + SIER_SHOW(RDR1_EN, rdr1); + SIER_SHOW(RDR0_EN, rdr0); + SIER_SHOW(TDE1_EN, tde1); + SIER_SHOW(TDE0_EN, tde0); + SIER_SHOW(ROE1_EN, roe1); + SIER_SHOW(ROE0_EN, roe0); + SIER_SHOW(TUE1_EN, tue1); + SIER_SHOW(TUE0_EN, tue0); + SIER_SHOW(TFS_EN, tfs); + SIER_SHOW(RFS_EN, rfs); + SIER_SHOW(TLS_EN, tls); + SIER_SHOW(RLS_EN, rls); + SIER_SHOW(RFF1_EN, rff1); + SIER_SHOW(RFF0_EN, rff0); + SIER_SHOW(TFE1_EN, tfe1); + SIER_SHOW(TFE0_EN, tfe0); return length; } diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d6882be33452..9c09b94f0cf8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,6 +146,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; + if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) { + /* + * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. + * Set constraint for minimum buffer size to the same than FIFO + * size in order to avoid underruns in playback startup because + * HW is keeping the DMA request active until FIFO is filled. + */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX); + } + if (!cpu_dai->active) err = omap_mcbsp_request(mcbsp_data->bus_id); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8e1431cb46bb..1bdbb0427183 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -327,7 +327,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, 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; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = omap_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 5998ab366e83..ad8a10fe6298 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_PXA2XX_SOC_MAGICIAN + tristate "SoC Audio support for HTC Magician" + depends on SND_PXA2XX_SOC && MACH_MAGICIAN + select SND_PXA2XX_SOC_I2S + select SND_PXA_SOC_SSP + select SND_SOC_UDA1380 + help + Say Y if you want to add support for SoC audio on the + HTC Magician. + config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 8ed881c5e5cc..4b90c3ccae45 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o @@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 000000000000..f7c4544f7859 --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,560 @@ +/* + * SoC audio for HTC Magician + * + * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com> + * + * based on spitz.c, + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> + * Richard Purdie <richard@openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/pxa-regs.h> +#include <mach/hardware.h> +#include <mach/magician.h> +#include <asm/mach-types.h> +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa-ssp.h" + +#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1 + +static int magician_hp_switch; +static int magician_spk_switch = 1; +static int magician_in_sel = MAGICIAN_MIC; + +static void magician_ext_control(struct snd_soc_codec *codec) +{ + if (magician_spk_switch) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + if (magician_hp_switch) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + switch (magician_in_sel) { + case MAGICIAN_MIC: + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case MAGICIAN_MIC_EXT: + snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + break; + } + + snd_soc_dapm_sync(codec); +} + +static int magician_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + /* check the jack status at stream startup */ + magician_ext_control(codec); + + return 0; +} + +/* + * Magician uses SSP port for playback. + */ +static int magician_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int acps, acds, width, rate; + unsigned int div4 = PXA_SSP_CLK_SCDB_4; + int ret = 0; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + + /* + * rate = SSPSCLK / (2 * width(16 or 32)) + * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) + */ + switch (params_rate(params)) { + case 8000: + /* off by a factor of 2: bug in the PXA27x audio clock? */ + acps = 32842000; + switch (width) { + case 16: + /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_16; + break; + case 32: + /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_8; + } + break; + case 11025: + acps = 5622000; + switch (width) { + case 16: + /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_4; + break; + case 32: + /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + } + break; + case 22050: + acps = 5622000; + switch (width) { + case 16: + /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 44100: + acps = 5622000; + switch (width) { + case 16: + /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 48000: + acps = 12235000; + switch (width) { + case 16: + /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 96000: + acps = 12235000; + switch (width) { + case 16: + /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + break; + case 32: + /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + div4 = PXA_SSP_CLK_SCDB_1; + break; + } + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + if (ret < 0) + return ret; + + /* set audio clock as clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set the SSP audio system clock ACDS divider */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + /* set the SSP audio system clock SCDB divider4 */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_SCDB, div4); + if (ret < 0) + return ret; + + /* set SSP audio pll clock */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Magician uses I2S for capture. + */ +static int magician_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops magician_capture_ops = { + .startup = magician_startup, + .hw_params = magician_capture_hw_params, +}; + +static struct snd_soc_ops magician_playback_ops = { + .startup = magician_startup, + .hw_params = magician_playback_hw_params, +}; + +static int magician_get_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_hp_switch; + return 0; +} + +static int magician_set_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_hp_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_hp_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_spk_switch; + return 0; +} + +static int magician_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_spk_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_spk_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_in_sel; + return 0; +} + +static int magician_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (magician_in_sel == ucontrol->value.integer.value[0]) + return 0; + + magician_in_sel = ucontrol->value.integer.value[0]; + + switch (magician_in_sel) { + case MAGICIAN_MIC: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); + break; + case MAGICIAN_MIC_EXT: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); + } + + return 1; +} + +static int magician_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* magician machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), + SND_SOC_DAPM_SPK("Speaker", magician_spk_power), + SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), + SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), +}; + +/* magician machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone connected to VOUTL, VOUTR */ + {"Headphone Jack", NULL, "VOUTL"}, + {"Headphone Jack", NULL, "VOUTR"}, + + /* Speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* Mics are connected to VINM */ + {"VINM", NULL, "Headset Mic"}, + {"VINM", NULL, "Call Mic"}, +}; + +static const char *input_select[] = {"Call Mic", "Headset Mic"}; +static const struct soc_enum magician_in_sel_enum = + SOC_ENUM_SINGLE_EXT(2, input_select); + +static const struct snd_kcontrol_new uda1380_magician_controls[] = { + SOC_SINGLE_BOOL_EXT("Headphone Switch", + (unsigned long)&magician_hp_switch, + magician_get_hp, magician_set_hp), + SOC_SINGLE_BOOL_EXT("Speaker Switch", + (unsigned long)&magician_spk_switch, + magician_get_spk, magician_set_spk), + SOC_ENUM_EXT("Input Select", magician_in_sel_enum, + magician_get_input, magician_set_input), +}; + +/* + * Logic for a uda1380 as connected on a HTC Magician + */ +static int magician_uda1380_init(struct snd_soc_codec *codec) +{ + int err; + + /* NC codec pins */ + snd_soc_dapm_nc_pin(codec, "VOUTLHP"); + snd_soc_dapm_nc_pin(codec, "VOUTRHP"); + + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(codec, "VINL"); + snd_soc_dapm_nc_pin(codec, "VINR"); + + /* Add magician specific controls */ + err = snd_soc_add_controls(codec, uda1380_magician_controls, + ARRAY_SIZE(uda1380_magician_controls)); + if (err < 0) + return err; + + /* Add magician specific widgets */ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + /* Set up magician specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* magician digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link magician_dai[] = { +{ + .name = "uda1380", + .stream_name = "UDA1380 Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], + .init = magician_uda1380_init, + .ops = &magician_playback_ops, +}, +{ + .name = "uda1380", + .stream_name = "UDA1380 Capture", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE], + .ops = &magician_capture_ops, +} +}; + +/* magician audio machine driver */ +static struct snd_soc_card snd_soc_card_magician = { + .name = "Magician", + .dai_link = magician_dai, + .num_links = ARRAY_SIZE(magician_dai), + .platform = &pxa2xx_soc_platform, +}; + +/* magician audio private data */ +static struct uda1380_setup_data magician_uda1380_setup = { + .i2c_address = 0x18, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +/* magician audio subsystem */ +static struct snd_soc_device magician_snd_devdata = { + .card = &snd_soc_card_magician, + .codec_dev = &soc_codec_dev_uda1380, + .codec_data = &magician_uda1380_setup, +}; + +static struct platform_device *magician_snd_device; + +static int __init magician_init(void) +{ + int ret; + + if (!machine_is_magician()) + return -ENODEV; + + ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); + if (ret) + goto err_request_power; + ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); + if (ret) + goto err_request_reset; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); + if (ret) + goto err_request_spk; + ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); + if (ret) + goto err_request_ep; + ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); + if (ret) + goto err_request_mic; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); + if (ret) + goto err_request_in_sel0; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); + if (ret) + goto err_request_in_sel1; + + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); + gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); + + /* we may need to have the clock running here - pH5 */ + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); + udelay(5); + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); + + magician_snd_device = platform_device_alloc("soc-audio", -1); + if (!magician_snd_device) { + ret = -ENOMEM; + goto err_pdev; + } + + platform_set_drvdata(magician_snd_device, &magician_snd_devdata); + magician_snd_devdata.dev = &magician_snd_device->dev; + ret = platform_device_add(magician_snd_device); + if (ret) { + platform_device_put(magician_snd_device); + goto err_pdev; + } + + return 0; + +err_pdev: + gpio_free(EGPIO_MAGICIAN_IN_SEL1); +err_request_in_sel1: + gpio_free(EGPIO_MAGICIAN_IN_SEL0); +err_request_in_sel0: + gpio_free(EGPIO_MAGICIAN_MIC_POWER); +err_request_mic: + gpio_free(EGPIO_MAGICIAN_EP_POWER); +err_request_ep: + gpio_free(EGPIO_MAGICIAN_SPK_POWER); +err_request_spk: + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); +err_request_reset: + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +err_request_power: + return ret; +} + +static void __exit magician_exit(void) +{ + platform_device_unregister(magician_snd_device); + + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); + + gpio_free(EGPIO_MAGICIAN_IN_SEL1); + gpio_free(EGPIO_MAGICIAN_IN_SEL0); + gpio_free(EGPIO_MAGICIAN_MIC_POWER); + gpio_free(EGPIO_MAGICIAN_EP_POWER); + gpio_free(EGPIO_MAGICIAN_SPK_POWER); + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +} + +module_init(magician_init); +module_exit(magician_exit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 7acd3febf8b0..308a657928d2 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) dma = 1; /* capture DMA offset is 1,3 */ - if (chn == 2) - dma += 2; /* stereo DMA offset is 2, mono is 0 */ + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + if (((chn == 2) && (ttsa != 1)) || (width == 32)) + dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); @@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !ttsa) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 53b9fb127a6d..d38e39575f51 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -81,7 +81,7 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { .mmap = pxa2xx_pcm_mmap, }; -static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -91,7 +91,7 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &pxa2xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index a9d68fa2b34a..169ddad31575 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -419,7 +419,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e710f705a74..99712f652d0d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = NULL; + codec->ac97->dev.parent = codec->card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", @@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - dev_dbg(socdev->dev, "scheduling resume work\n"); - - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + /* AC97 devices might have other drivers hanging off them so + * need to resume immediately. Other drivers don't have that + * problem and may take a substantial amount of time to resume + * due to I/O costs and anti-pop so handle them out of line. + */ + if (cpu_dai->ac97_control) { + dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(socdev->dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(socdev->dev, "resume work item may be lost\n"); + } return 0; } diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 7d93fa705ccf..8d13d933087d 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -703,7 +703,7 @@ static int snd_cs4231_timer_stop(struct snd_timer *timer) return 0; } -static void __init snd_cs4231_init(struct snd_cs4231 *chip) +static void __devinit snd_cs4231_init(struct snd_cs4231 *chip) { unsigned long flags; @@ -1020,7 +1020,7 @@ static snd_pcm_uframes_t snd_cs4231_capture_pointer( return bytes_to_frames(substream->runtime, ptr); } -static int __init snd_cs4231_probe(struct snd_cs4231 *chip) +static int __devinit snd_cs4231_probe(struct snd_cs4231 *chip) { unsigned long flags; int i; @@ -1219,7 +1219,7 @@ static struct snd_pcm_ops snd_cs4231_capture_ops = { .pointer = snd_cs4231_capture_pointer, }; -static int __init snd_cs4231_pcm(struct snd_card *card) +static int __devinit snd_cs4231_pcm(struct snd_card *card) { struct snd_cs4231 *chip = card->private_data; struct snd_pcm *pcm; @@ -1248,7 +1248,7 @@ static int __init snd_cs4231_pcm(struct snd_card *card) return 0; } -static int __init snd_cs4231_timer(struct snd_card *card) +static int __devinit snd_cs4231_timer(struct snd_card *card) { struct snd_cs4231 *chip = card->private_data; struct snd_timer *timer; @@ -1499,7 +1499,7 @@ static int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, .private_value = (left_reg) | ((right_reg) << 8) | ((shift_left) << 16) | \ ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22) } -static struct snd_kcontrol_new snd_cs4231_controls[] __initdata = { +static struct snd_kcontrol_new snd_cs4231_controls[] __devinitdata = { CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, @@ -1538,7 +1538,7 @@ CS4231_SINGLE("Line Out Switch", 0, CS4231_PIN_CTRL, 6, 1, 1), CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1) }; -static int __init snd_cs4231_mixer(struct snd_card *card) +static int __devinit snd_cs4231_mixer(struct snd_card *card) { struct snd_cs4231 *chip = card->private_data; int err, idx; @@ -1559,7 +1559,7 @@ static int __init snd_cs4231_mixer(struct snd_card *card) static int dev; -static int __init cs4231_attach_begin(struct snd_card **rcard) +static int __devinit cs4231_attach_begin(struct snd_card **rcard) { struct snd_card *card; struct snd_cs4231 *chip; @@ -1590,7 +1590,7 @@ static int __init cs4231_attach_begin(struct snd_card **rcard) return 0; } -static int __init cs4231_attach_finish(struct snd_card *card) +static int __devinit cs4231_attach_finish(struct snd_card *card) { struct snd_cs4231 *chip = card->private_data; int err; @@ -1794,9 +1794,9 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = { .dev_free = snd_cs4231_sbus_dev_free, }; -static int __init snd_cs4231_sbus_create(struct snd_card *card, - struct of_device *op, - int dev) +static int __devinit snd_cs4231_sbus_create(struct snd_card *card, + struct of_device *op, + int dev) { struct snd_cs4231 *chip = card->private_data; int err; @@ -1960,9 +1960,9 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = { .dev_free = snd_cs4231_ebus_dev_free, }; -static int __init snd_cs4231_ebus_create(struct snd_card *card, - struct of_device *op, - int dev) +static int __devinit snd_cs4231_ebus_create(struct snd_card *card, + struct of_device *op, + int dev) { struct snd_cs4231 *chip = card->private_data; int err; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c2db0f959681..823296d7d578 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -121,6 +121,7 @@ struct audioformat { unsigned char attributes; /* corresponding attributes of cs endpoint */ unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ + unsigned char datainterval; /* log_2 of data packet interval */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ @@ -170,7 +171,6 @@ struct snd_usb_substream { unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int fmt_type; /* USB audio format type (1-3) */ - unsigned int packs_per_ms; /* packets per millisecond (for playback) */ unsigned int running: 1; /* running status */ @@ -607,9 +607,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, break; } } - /* finish at the frame boundary at/after the period boundary */ - if (period_elapsed && - (i & (subs->packs_per_ms - 1)) == subs->packs_per_ms - 1) + if (period_elapsed) /* finish at the period boundary */ break; } if (subs->hwptr_done + offs > runtime->buffer_size) { @@ -1067,7 +1065,6 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; - subs->packs_per_ms = packs_per_ms; if (is_playback) { urb_packs = max(nrpacks, 1); @@ -1087,18 +1084,17 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri minsize -= minsize >> 3; minsize = max(minsize, 1u); total_packs = (period_bytes + minsize - 1) / minsize; - /* round up to multiple of packs_per_ms */ - total_packs = (total_packs + packs_per_ms - 1) - & ~(packs_per_ms - 1); /* we need at least two URBs for queueing */ - if (total_packs < 2 * packs_per_ms) { - total_packs = 2 * packs_per_ms; + if (total_packs < 2) { + total_packs = 2; } else { /* and we don't want too long a queue either */ maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); total_packs = min(total_packs, maxpacks); } } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; total_packs = MAX_URBS * urb_packs; } subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; @@ -1350,12 +1346,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datapipe = usb_sndisocpipe(dev, ep); else subs->datapipe = usb_rcvisocpipe(dev, ep); - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - subs->datainterval = get_endpoint(alts, 0)->bInterval - 1; - else - subs->datainterval = 0; + subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; subs->fill_max = 0; @@ -1568,11 +1559,15 @@ static struct snd_pcm_hardware snd_usb_hardware = #define hwc_debug(fmt, args...) /**/ #endif -static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audioformat *fp) +static int hw_check_valid_format(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *params, + struct audioformat *fp) { struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + unsigned int ptime; /* check the format */ if (!snd_mask_test(fmts, fp->format)) { @@ -1593,6 +1588,14 @@ static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audiof hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min); return 0; } + /* check whether the period time is >= the data packet interval */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + ptime = 125 * (1 << fp->datainterval); + if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { + hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); + return 0; + } + } return 1; } @@ -1611,7 +1614,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->rate_min) @@ -1665,7 +1668,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->channels) @@ -1718,7 +1721,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; fbits |= (1ULL << fp->format); } @@ -1736,95 +1739,42 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } -#define MAX_MASK 64 - -/* - * check whether the registered audio formats need special hw-constraints - */ -static int check_hw_params_convention(struct snd_usb_substream *subs) +static int hw_rule_period_time(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { - int i; - u32 *channels; - u32 *rates; - u32 cmaster, rmaster; - u32 rate_min = 0, rate_max = 0; - struct list_head *p; - int err = 1; - - channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - if (!channels || !rates) { - err = -ENOMEM; - goto __out; - } + struct snd_usb_substream *subs = rule->private; + struct audioformat *fp; + struct snd_interval *it; + unsigned char min_datainterval; + unsigned int pmin; + int changed; - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - /* unconventional channels? */ - if (f->channels > 32) - goto __out; - /* continuous rate min/max matches? */ - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rate_min && f->rate_min != rate_min) - goto __out; - if (rate_max && f->rate_max != rate_max) - goto __out; - rate_min = f->rate_min; - rate_max = f->rate_max; - } - /* combination of continuous rates and fixed rates? */ - if (rates[f->format] & SNDRV_PCM_RATE_CONTINUOUS) { - if (f->rates != rates[f->format]) - goto __out; - } - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rates[f->format] && rates[f->format] != f->rates) - goto __out; - } - channels[f->format] |= 1 << (f->channels - 1); - rates[f->format] |= f->rates; - /* needs knot? */ - if (f->rates & SNDRV_PCM_RATE_KNOT) - goto __out; - } - /* check whether channels and rates match for all formats */ - cmaster = rmaster = 0; - for (i = 0; i < MAX_MASK; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (rmaster != rates[i] && rmaster && rates[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; - if (rates[i]) - rmaster = rates[i]; - } - /* check whether channels match for all distinct rates */ - memset(channels, 0, MAX_MASK * sizeof(u32)); - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max); + min_datainterval = 0xff; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) continue; - for (i = 0; i < 32; i++) { - if (f->rates & (1 << i)) - channels[i] |= 1 << (f->channels - 1); - } + min_datainterval = min(min_datainterval, fp->datainterval); } - cmaster = 0; - for (i = 0; i < 32; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; + if (min_datainterval == 0xff) { + hwc_debug(" --> get emtpy\n"); + it->empty = 1; + return -EINVAL; } - err = 0; - - __out: - kfree(channels); - kfree(rates); - return err; + pmin = 125 * (1 << min_datainterval); + changed = 0; + if (it->min < pmin) { + it->min = pmin; + it->openmin = 0; + changed = 1; + } + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed); + return changed; } /* @@ -1872,6 +1822,8 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { struct list_head *p; + unsigned int pt, ptmin; + int param_period_time_if_needed; int err; runtime->hw.formats = subs->formats; @@ -1881,6 +1833,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = 256; runtime->hw.channels_max = 0; runtime->hw.rates = 0; + ptmin = UINT_MAX; /* check min/max rates and channels */ list_for_each(p, &subs->fmt_list) { struct audioformat *fp; @@ -1899,42 +1852,54 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; } + pt = 125 * (1 << fp->datainterval); + ptmin = min(ptmin, pt); } - /* set the period time minimum 1ms */ - /* FIXME: high-speed mode allows 125us minimum period, but many parts - * in the current code assume the 1ms period. - */ + param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; + if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + /* full speed devices have fixed data packet interval */ + ptmin = 1000; + if (ptmin == 1000) + /* if period time doesn't go below 1 ms, no rules needed */ + param_period_time_if_needed = -1; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000, - /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); - - err = check_hw_params_convention(subs); - if (err < 0) + ptmin, UINT_MAX); + + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) return err; - else if (err) { - hwc_debug("setting extra hw constraints...\n"); - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) + return err; + if (param_period_time_if_needed >= 0) { + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + hw_rule_period_time, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, + -1); + if (err < 0) return err; } + if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) + return err; return 0; } @@ -2147,7 +2112,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %#x\n", fp->format); + snd_iprintf(buffer, " Format: %#x (%d bits)\n", + fp->format, snd_pcm_format_width(fp->format)); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, @@ -2166,6 +2132,9 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + snd_iprintf(buffer, " Data packet interval: %d us\n", + 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } @@ -2659,6 +2628,17 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp return 0; } +static unsigned char parse_datainterval(struct snd_usb_audio *chip, + struct usb_host_interface *alts) +{ + if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && + get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + else + return 0; +} + static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) @@ -2764,6 +2744,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->altset_idx = i; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) @@ -2955,6 +2936,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); @@ -3049,6 +3031,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = 0; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); switch (fp->maxpacksize) { @@ -3116,6 +3099,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]); @@ -3168,6 +3152,7 @@ static int create_ua101_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); 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