*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
- if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
- BUG();
- if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
- BUG();
+ BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+ (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
module_init(n810_soc_init);
module_exit(n810_soc_exit);
- MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+ MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
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;
- int wlen, channels;
+ int wlen, channels, wpf;
unsigned long port;
+ unsigned int format;
if (cpu_class_is_omap1()) {
dma = omap1_dma_reqs[bus_id][substream->stream];
return 0;
}
- channels = params_channels(params);
+ format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+ wpf = channels = params_channels(params);
switch (channels) {
case 2:
- /* Use dual-phase frames */
- regs->rcr2 |= RPHASE;
- regs->xcr2 |= XPHASE;
+ if (format == SND_SOC_DAIFMT_I2S) {
+ /* Use dual-phase frames */
+ regs->rcr2 |= RPHASE;
+ regs->xcr2 |= XPHASE;
+ /* Set 1 word per (McBSP) frame for phase1 and phase2 */
+ wpf--;
+ regs->rcr2 |= RFRLEN2(wpf - 1);
+ regs->xcr2 |= XFRLEN2(wpf - 1);
+ }
case 1:
- /* Set 1 word per (McBSP) frame */
- regs->rcr2 |= RFRLEN2(1 - 1);
- regs->rcr1 |= RFRLEN1(1 - 1);
- regs->xcr2 |= XFRLEN2(1 - 1);
- regs->xcr1 |= XFRLEN1(1 - 1);
+ /* Set word per (McBSP) frame for phase1 */
+ regs->rcr1 |= RFRLEN1(wpf - 1);
+ regs->xcr1 |= XFRLEN1(wpf - 1);
break;
default:
/* Unsupported number of channels */
}
/* Set FS period and length in terms of bit clock periods */
- switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ switch (format) {
case SND_SOC_DAIFMT_I2S:
- regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(wlen - 1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(0);
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* 1-bit data delay */
+ regs->rcr2 |= RDATDLY(1);
+ regs->xcr2 |= XDATDLY(1);
+ /* Invert FS polarity configuration */
+ temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+ break;
case SND_SOC_DAIFMT_DSP_B:
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
}
module_exit(snd_omap_mcbsp_exit);
- MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+ MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP I2S SoC Interface");
MODULE_LICENSE("GPL");
*
* Copyright (C) 2008 Nokia Corporation
*
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ * Peter Ujfalusi <peter.ujfalusi@nokia.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
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,
}
module_exit(omap_soc_platform_exit);
- MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+ MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
MODULE_DESCRIPTION("OMAP PCM DMA module");
MODULE_LICENSE("GPL");
#include <sound/pxa2xx-lib.h>
#include <mach/hardware.h>
-#include <mach/pxa-regs.h>
+#include <mach/dma.h>
#include <mach/regs-ssp.h>
#include <mach/audio.h>
#include <mach/ssp.h>
* ssp_set_clkdiv - set SSP clock divider
* @div: serial clock rate divider
*/
- static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+ static void ssp_set_scr(struct ssp_device *ssp, u32 div)
{
- struct ssp_device *ssp = dev->ssp;
- u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+ sscr0 &= ~0x0000ff00;
+ sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+ } else {
+ sscr0 &= ~0x000fff00;
+ sscr0 |= (div - 1) << 8; /* 1..4096 */
+ }
+ ssp_write_reg(ssp, SSCR0, sscr0);
+ }
+
+ /**
+ * ssp_get_clkdiv - get SSP clock divider
+ */
+ static u32 ssp_get_scr(struct ssp_device *ssp)
+ {
+ u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+ u32 div;
- ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+ if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+ div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+ else
+ div = ((sscr0 >> 8) & 0xfff) + 1;
+ return div;
}
/*
break;
case PXA_SSP_CLK_AUDIO:
priv->sysclk = 0;
- ssp_set_scr(&priv->dev, 1);
+ ssp_set_scr(ssp, 1);
sscr0 |= SSCR0_ACS;
break;
default:
ssp_write_reg(ssp, SSACD, val);
break;
case PXA_SSP_DIV_SCR:
- ssp_set_scr(&priv->dev, div);
+ ssp_set_scr(ssp, div);
break;
default:
return -ENODEV;
case SND_SOC_DAIFMT_NB_IF:
break;
case SND_SOC_DAIFMT_IB_IF:
- sspsp |= SSPSP_SCMODE(3);
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
break;
default:
return -EINVAL;
case SND_SOC_DAIFMT_NB_NF:
sspsp |= SSPSP_SFRMP;
break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
case SND_SOC_DAIFMT_IB_IF:
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
break;
default:
return -EINVAL;
case SND_SOC_DAIFMT_I2S:
sspsp = ssp_read_reg(ssp, SSPSP);
- if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
- (width == 16)) {
+ if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
/* This is a special case where the bitclk is 64fs
* and we're not dealing with 2*32 bits of audio
* samples.