From 236cc52856f6ebe47f52d50ba5431b0e172fd0d1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Sep 2009 12:46:42 +0100 Subject: ASoC: Remove unuused hw_read_t Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 0758a1b..475cb7e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -205,7 +205,6 @@ struct snd_soc_jack_gpio; #endif typedef int (*hw_write_t)(void *,const char* ,int); -typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; -- cgit v0.10.2 From 87831cb660954356d68cebdb1406f3be09e784e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Sep 2009 18:09:58 +0100 Subject: ASoC: Fix WM835x Out4 capture enumeration It's the 8th enum of a zero indexed array. This is why I don't let new drivers use these arrays of enums... Signed-off-by: Mark Brown Cc: stable@kernel.org diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 71c9c4b..3ff0373 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -612,7 +612,7 @@ SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1); /* Out4 Capture Mux */ static const struct snd_kcontrol_new wm8350_out4_capture_controls = -SOC_DAPM_ENUM("Route", wm8350_enum[8]); +SOC_DAPM_ENUM("Route", wm8350_enum[7]); static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = { -- cgit v0.10.2 From cdc65fbe18aef15e92d2ebb410a189fbf956fb06 Mon Sep 17 00:00:00 2001 From: Manuel Lauss Date: Tue, 8 Sep 2009 19:45:17 +0200 Subject: ASoC: au1x: PSC-AC97 bugfixes This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss Signed-off-by: Mark Brown diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss "); +MODULE_AUTHOR("Manuel Lauss "); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a04..3f474e8 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 -- cgit v0.10.2 From 33d7f77850476a8b8df50bd50221bc644dd44357 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 12 Sep 2009 14:25:35 +0200 Subject: ASoC: Clean up error handling in MPC5200 DMA setup Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // Signed-off-by: Julia Lawall Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..6096d22 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op) int size, irq, rc; const __be32 *prop; void __iomem *regs; + int ret; /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); @@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { - iounmap(regs); - return -ENOMEM; + ret = -ENOMEM; + goto out_unmap; } /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; + if (!prop || size < sizeof *prop) { + ret = -ENODEV; + goto out_free; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex); @@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op) if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + ret = -ENODEV; + goto out_free; } /* Disable all interrupts and reset the PSC */ @@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op) &psc_dma_bcom_irq_tx, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; + ret = -ENODEV; + goto out_irq; } /* Save what we've done so it can be found again later */ @@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Tell the ASoC OF helpers about it */ return snd_soc_register_platform(&mpc5200_audio_dma_platform); +out_irq: + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); +out_free: + kfree(psc_dma); +out_unmap: + iounmap(regs); + return ret; } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); -- cgit v0.10.2 From 3eef08ba522775360cc59fe0a6b1bca6ecc8da4e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 14 Sep 2009 16:49:00 +0100 Subject: ASoC: Fix display of stream name in DAPM debugfs Also display streams all the time while we're here. Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8b08e..f79711b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1131,9 +1131,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file, ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n", w->name, w->power ? "On" : "Off", in, out); - if (w->active && w->sname) - ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n", - w->sname); + if (w->sname) + ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + w->sname, + w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { if (p->connect) -- cgit v0.10.2 From fa68e0025d4184ba917621a9c977d4243d0a013e Mon Sep 17 00:00:00 2001 From: Jassi Date: Tue, 15 Sep 2009 19:02:37 +0900 Subject: ASoC: S3C lrsync function made to work with IRQs disabled. s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK is dead due to improper initialization of CPU or CODEC, the system gets stuck in the loop because jiffies may never get updated. Implemented counter based wait mechanism for atleast the same timeout period. Signed-off-by: Jassi Signed-off-by: Mark Brown diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aa7af0b..9bc4aa3 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + /* * Wait for the LR signal to allow synchronisation to the L/R clock * from the codec. May only be needed for slave mode. @@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + unsigned long loops = msecs_to_loops(5); pr_debug("Entered %s\n", __func__); - while (1) { + while (--loops) { iiscon = readl(i2s->regs + S3C2412_IISCON); if (iiscon & S3C2412_IISCON_LRINDEX) break; - if (timeout < jiffies) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; } return 0; -- cgit v0.10.2 From d4e54e871f4d2ca29df081abf8e0d5209d252979 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Wed, 16 Sep 2009 21:05:45 +0800 Subject: ASoC: remove unused #include Remove unused #include ('s) in sound/soc/codecs/ad1836.c sound/soc/codecs/ad1938.c sound/soc/codecs/wm8974.c Signed-off-by: Huang Weiyi Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3612bb9..01343dc 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -18,7 +18,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index e62b277..9a049a1 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -28,7 +28,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d8a013a..98d663a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -12,7 +12,6 @@ #include #include -#include #include #include #include -- cgit v0.10.2 From 79dfc9687661c13ef95eb4c2226f3db4ccab52c9 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:08 -0400 Subject: ASoC: Blackfin AC97: add a few missing multichannel define handling Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was merged, but two places were missed (the probe/resume functions). Restore handling of this option so it gets initialized properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b90..e693229 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; -- cgit v0.10.2 From d75150d7c49db42021b8f966d2cbdc215a530208 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Wed, 16 Sep 2009 20:25:09 -0400 Subject: ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 469ce7f..99051ff 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) int sport_tx_start(struct sport_device *sport) { - unsigned flags; + unsigned long flags; pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, sport->tx_run, sport->rx_run); if (sport->tx_run) -- cgit v0.10.2 From 7d156a25bd3e8e6ff74faf02faecb5fc5fb4839e Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:10 -0400 Subject: ASoC: fix typos in Blackfin headers Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f2a911..a1f97dd 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-ac97.h + * sound/soc/blackfin/bf5xx-ac97.h * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h index 7107d1a..264ecdc 100644 --- a/sound/soc/blackfin/bf5xx-i2s.h +++ b/sound/soc/blackfin/bf5xx-i2s.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-i2s.h * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as -- cgit v0.10.2 From fab19bae0c2951ed8bc517a53848b027fead293d Mon Sep 17 00:00:00 2001 From: Barry Song Date: Wed, 16 Sep 2009 20:25:11 -0400 Subject: ASoC: Blackfin I2S: add lost platform_device parameter to resume function Commit dc7d7b830ee1 trimmed the platform_device parameter from all of the suspend functions, but it also accidentally removed it from the resume function in the Blackfin I2S driver. So restore it. Signed-off-by: Barry Song Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abad..19539c68 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } -static void bf5xx_i2s_remove(struct snd_soc_dai *dai) +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); peripheral_free_list(&sport_req[sport_num][0]); -- cgit v0.10.2 From ad80efc469f56d41f3f4adc1b2c86bf65689ebeb Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Wed, 16 Sep 2009 20:25:12 -0400 Subject: ASoC: Blackfin I2S: fix resuming when device hasn't been used If the sound system hasn't been utilized yet and we suspend, then we attempt to save/restore using state that doesn't exist. So use a global handle instead to reconfigure properly. Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 19539c68..1e9d161 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -237,36 +237,31 @@ static void bf5xx_i2s_remove(struct platform_device *pdev, #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; + if (dai->capture.active) - sport_rx_stop(sport); + sport_rx_stop(sport_handle); if (dai->playback.active) - sport_tx_stop(sport); + sport_tx_stop(sport_handle); return 0; } static int bf5xx_i2s_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; -- cgit v0.10.2 From 0c31cf3e4af79ea18bbd365b07ef0de207673894 Mon Sep 17 00:00:00 2001 From: Chaithrika U S Date: Tue, 15 Sep 2009 18:13:29 -0400 Subject: ASoC: DaVinci: Fixes to McASP configuration McASP register settings are not correct for DSP mode of operation. There is a channel swap initally. This patch provides fixes to the register values for proper working. Tested on DA830/OMAP-L137 EVM, DM6467 EVM. Signed-off-by: Chaithrika U S Signed-off-by: Mark Brown diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7..7a06c0a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -512,34 +512,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +565,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, RXSSZ(fmt), RXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + return 0; } @@ -638,7 +660,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +676,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } -- cgit v0.10.2