diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-04-19 10:57:38 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-04-19 10:57:38 -0700 |
commit | af8f937274437fa81b95e4e2d461460220636cb8 (patch) | |
tree | a0fce546e4693e759ed944ba37603c36bf514430 /sound/soc | |
parent | 091ccb006fcf5c4aa1283901ca6e62ff85b3a569 (diff) | |
parent | d6aa764ee8674512287913fcc3a0b1b5c050d5eb (diff) |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Set function_id only on FG nodes
ALSA: MAINTAINERS - Update SOUND
ALSA: emu10k1 - off by 1 in snd_emu10k1_wait()
ASoC: OMAP: Fix FS polarity in OSK5912 machine driver
ASoC: OMAP: Fix DSP_B format in OMAP McBSP DAI driver
ASoC: Fix include build error in s3c2412-i2s.c
ASoC: Fix s3c-i2s-v2.c snd_soc_dai changes
ASoC: s3c-i2s-v2.c fix for s3c_i2sv2_iis_calc_rate
ASoC: Fix jive_wm8750.c build problems
ASoC: pxa-ssp: allow setting of dai format 0
ALSA: hda - Add upper-limit of mixer amp for AD1884A-laptop model, too
ALSA: hda - Fix headphone-detection on some machines with STAC/IDT codecs
ALSA: Intel8x0: Add hp_only quirk for SSID 0x1028016a (Dell Inspiron 8600)
ALSA: Intel8x0: Remove conflicting quirk for SSID 0x103c0934
ALSA: hda_intel.c - Consolidate bitfields
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/omap/omap-mcbsp.c | 7 | ||||
-rw-r--r-- | sound/soc/omap/osk5912.c | 4 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 1 | ||||
-rw-r--r-- | sound/soc/s3c24xx/jive_wm8750.c | 12 | ||||
-rw-r--r-- | sound/soc/s3c24xx/s3c-i2s-v2.c | 18 | ||||
-rw-r--r-- | sound/soc/s3c24xx/s3c2412-i2s.c | 2 |
6 files changed, 25 insertions, 19 deletions
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9c09b94f0cf..90f4df7fd90 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -283,7 +283,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen * channels - 2); + regs->srgr1 |= FWID(0); break; } @@ -302,6 +302,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -328,6 +329,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; break; default: /* Unsupported data format */ @@ -351,7 +354,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index a952a4eb336..a4e149b7f0e 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set codec DAI configuration */ err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set codec DAI configuration\n"); @@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set cpu DAI configuration */ err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657928d..de2254475d5 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -806,6 +806,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, goto err_priv; } + priv->dai_fmt = (unsigned int) -1; dai->private_data = priv; return 0; diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 32063790d95..93e6c87b739 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, break; } - s3c_i2sv2_calc_rate(&div, NULL, params_rate(params), - s3c2412_get_iisclk()); + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | @@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = { }; /* jive audio machine driver */ -static struct snd_soc_machine snd_soc_machine_jive = { +static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", + .platform = &s3c24xx_soc_platform, .dai_link = &jive_dai, .num_links = 1, }; @@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = { /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { - .machine = &snd_soc_machine_jive, - .platform = &s3c24xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8750_spi, + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, .codec_data = &jive_wm8750_setup, }; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 295a4c91026..689ffcd17e1 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; -int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) { unsigned long clkrate = clk_get_rate(clk); unsigned int div; @@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, return 0; } -EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct platform_device *pdev, struct snd_soc_dai *dai, @@ -624,10 +624,12 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) { - dai->ops.trigger = s3c2412_i2s_trigger; - dai->ops.hw_params = s3c2412_i2s_hw_params; - dai->ops.set_fmt = s3c2412_i2s_set_fmt; - dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv; + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + ops->hw_params = s3c2412_i2s_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 1ca3cdaa821..b7e0b3f0bfc 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -33,8 +33,8 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/regs-gpio.h> #include <plat/audio.h> +#include <mach/regs-gpio.h> #include <mach/dma.h> #include "s3c24xx-pcm.h" |