From 5e901b37e4a8a305542ad3a776bce997efd7e5e9 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Thu, 9 Apr 2009 14:07:27 +0800 Subject: [ARM] pxa/magician: remove un-necessary #include of pxa-regs.h and hardware.h Signed-off-by: Eric Miao Cc: Philipp Zabel Cc: Mark Brown --- sound/soc/pxa/magician.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index f7c4544f7859..0625c342a1c9 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -27,8 +27,6 @@ #include #include -#include -#include #include #include #include "../codecs/uda1380.h" -- cgit v1.2.2 From 1a297286868e13274ab02ec0626a00054fb0a5de Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Fri, 17 Apr 2009 11:39:38 +0200 Subject: ASoC: pxa-ssp: Don't use SSCR0_SerClkDiv and SSCR0_SCR Those macros are just screwed as soon as CONFIG_PXA25x is enabled. This patch - changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device - adds a corresponding ssp_get_scr function. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 36 ++++++++++++++++++++++++++++-------- 1 file changed, 28 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 152118cb5d61..74ff69e3ce34 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * 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; } /* @@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, 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: @@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, 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; @@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, 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. -- cgit v1.2.2 From b08f7a62cafd7998862072a1c353219e3d84bbef Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 17 Apr 2009 14:42:26 +0300 Subject: ASoC: OMAP: Update contact addresses My email address is going to expire soon so update it. Adding also Peter Ujfalusi as a second contact to OMAP core drivers since I won't have anymore access to non-public OMAP documentation in the future and Peter is working with these drivers as well. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 5 +++-- sound/soc/omap/omap-mcbsp.h | 3 ++- sound/soc/omap/omap-pcm.c | 5 +++-- sound/soc/omap/omap-pcm.h | 3 ++- 5 files changed, 12 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index a6d1178ce128..91ef17992de5 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void) module_init(n810_soc_init); module_exit(n810_soc_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 90f4df7fd906..912614283848 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -532,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void) } module_exit(snd_omap_mcbsp_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index df7ad13ba73d..c8147aace813 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8e1431cb46bb..b078ed537bc3 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void) } module_exit(omap_soc_platform_exit); -MODULE_AUTHOR("Jarkko Nikula "); +MODULE_AUTHOR("Jarkko Nikula "); MODULE_DESCRIPTION("OMAP PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index e4369bdfd77d..8d9d26916b05 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -3,7 +3,8 @@ * * Copyright (C) 2008 Nokia Corporation * - * Contact: Jarkko Nikula + * Contact: Jarkko Nikula + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- cgit v1.2.2 From 516ef69f160fb2f092d71f2cb635a9394ff8a71a Mon Sep 17 00:00:00 2001 From: Russell King - ARM Linux Date: Sat, 18 Apr 2009 10:11:53 +0100 Subject: ASoC: Fix warning in wm9705 I notice that the fixes were merged, minus one: sound/soc/codecs/wm9705.c: At top level: sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type so you might find this trivial patch useful. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 6e23a81dba78..c2d1a7a18fa3 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int wm9705_soc_suspend(struct platform_device *pdev) +static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; -- cgit v1.2.2 From e91fb9137dd235ab959d7675d0e4104974dad5eb Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Fri, 17 Apr 2009 11:37:35 +0200 Subject: [ARM] pxa/palm27x: General fix for Palm27x aSoC driver Firstly, this patch makes the palm27x asoc driver a little more sane. Also, since all affected devices use GPIO95 as AC97_nRESET, this patch sets that properly. Affected are PalmT5, TX and LifeDrive. Signed-off-by: Marek Vasut Signed-off-by: Eric Miao --- sound/soc/pxa/palm27x.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 48a73f64500b..44fcc4e01e08 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = { static struct platform_device *palm27x_snd_device; -static int __init palm27x_asoc_init(void) +static int palm27x_asoc_probe(struct platform_device *pdev) { int ret; @@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void) machine_is_palmld())) return -ENODEV; + if (pdev->dev.platform_data) + palm27x_ep_gpio = ((struct palm27x_asoc_info *) + (pdev->dev.platform_data))->jack_gpio; + ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); if (ret) return ret; @@ -245,16 +249,31 @@ err_alloc: return ret; } -static void __exit palm27x_asoc_exit(void) +static int __devexit palm27x_asoc_remove(struct platform_device *pdev) { free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); gpio_free(palm27x_ep_gpio); platform_device_unregister(palm27x_snd_device); + return 0; } -void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) +static struct platform_driver palm27x_wm9712_driver = { + .probe = palm27x_asoc_probe, + .remove = __devexit_p(palm27x_asoc_remove), + .driver = { + .name = "palm27x-asoc", + .owner = THIS_MODULE, + }, +}; + +static int __init palm27x_asoc_init(void) +{ + return platform_driver_register(&palm27x_wm9712_driver); +} + +static void __exit palm27x_asoc_exit(void) { - palm27x_ep_gpio = data->jack_gpio; + platform_driver_unregister(&palm27x_wm9712_driver); } module_init(palm27x_asoc_init); -- cgit v1.2.2 From ce88168f5b5eca7f40394fa6b05ae29f4b685569 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Apr 2009 12:35:15 +0100 Subject: ASoC: Fix offset of freqmode in WM8580 PLL configuration Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 442ea6f160fc..41aab4a7a254 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -522,7 +522,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, reg = wm8580_read(codec, WM8580_PLLA4 + offset); reg &= ~0x3f; reg |= pll_div.prescale | pll_div.postscale << 1 | - pll_div.freqmode << 4; + pll_div.freqmode << 3; wm8580_write(codec, WM8580_PLLA4 + offset, reg); -- cgit v1.2.2 From ccb077fd466ec3f35662d6c66412b42b36e11bc3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Apr 2009 12:57:00 +0100 Subject: ASoC: Fix WM8580 volume update handling for large register changes The driver is out of sync with the core functions it is using. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 41aab4a7a254..9f6be3d31ac0 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int reg2 = (kcontrol->private_value >> 24) & 0xff; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; int ret; u16 val; @@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, return 0; } -#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \ +#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ SNDRV_CTL_ELEM_ACCESS_READWRITE, \ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw_2r, \ .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \ - .private_value = (reg_left) | ((shift) << 8) | \ - ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } static const struct snd_kcontrol_new wm8580_snd_controls[] = { SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume", -- cgit v1.2.2 From a1992db55d80297544a65160ddb98afba45f7759 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 23 Apr 2009 08:22:23 +0200 Subject: ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991 Signed-off-by: Takashi Iwai --- sound/soc/codecs/Makefile | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d2454725f..f2653803ede8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,7 +56,6 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o -obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o -- cgit v1.2.2 From a396e32ef0f3b98700abb9a6da3530c945e55908 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Apr 2009 15:43:45 +0100 Subject: ASoC: s3c-i2s-v2 needs to declare a license for modular builds It relies on EXPORT_SYMBOL_GPL() symbols. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 689ffcd17e1f..ab680aac3fcb 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -636,5 +636,6 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) return snd_soc_register_dai(dai); } - EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); -- cgit v1.2.2 From 5e42336a461a2354b640001323cd07cebd9ade6e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Apr 2009 19:18:22 +0100 Subject: ASoC: Fix logic in WM8350 master clocking check We need to check only if the WM8350 is master and only when starting the stream so if either is not true then we can skip the check. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b1d0993bed9..0275321ff8ab 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, * required for LRC in master mode. The DACs or ADCs need a * valid audio path i.e. pin -> ADC or DAC -> pin before * the LRC will be enabled in master mode. */ - if (!master && cmd != SNDRV_PCM_TRIGGER_START) + if (!master || cmd != SNDRV_PCM_TRIGGER_START) return 0; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { -- cgit v1.2.2 From 0c95de73a711d376dc17afe484f919bd5b66c016 Mon Sep 17 00:00:00 2001 From: Jon Smirl Date: Mon, 27 Apr 2009 12:44:41 -0400 Subject: ASoC: Set the MPC5200 i2s driver to BROKEN status. Signed-off-by: Jon Smirl Acked-by: Grant Likely Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 9fc908283371..e7dd79a1d8cd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -21,7 +21,7 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM + depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN select SND_SOC_OF_SIMPLE select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.2 From 2008f137e92220b98120c4803499cdddb2b0fb06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Apr 2009 12:25:59 +0200 Subject: ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some drivers Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers that really don't give the precise pointer value. Signed-off-by: Takashi Iwai --- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 3 ++- sound/soc/sh/dma-sh7760.c | 3 ++- 3 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 30490a259148..594c6c5b7838 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2]; /* PCM hardware DMA capabilities - platform specific */ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH, .formats = AU1XPSC_PCM_FMTS, .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES, .period_bytes_max = 4096 * 1024 - 1, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 3aa729df27b5..1111c710118a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = { static const struct snd_pcm_hardware psc_i2s_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_BATCH, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, .rate_min = 8000, diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 0dad3a0bb920..baddb1242c71 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH), .formats = DMABRG_FMTS, .rates = DMABRG_RATES, .rate_min = 8000, -- cgit v1.2.2 From 18cc8d8d9b74c446832336d8f6e1afb145f9431b Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 28 Apr 2009 18:18:05 +0900 Subject: ASoC: TWL4030: Fix gain control for earpiece amplifier The gain control for earpiece amplifier uses 0dB ~ 12dB according to the TRM, but the present code is implemented to -6dB ~ 6dB. Signed-off-by: Joonyoung Shim Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205de28a..df7c8c281d2f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -835,6 +835,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); */ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); +/* + * Gain control for earpiece amplifier + * 0 dB to 12 dB in 6 dB steps (mute instead of -6) + */ +static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1); + /* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps @@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 4, 3, 0, output_tvl), SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", - TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume", -- cgit v1.2.2 From 6574612fbb34c63117581e68f2231ddce027e41e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 May 2009 16:03:21 +0200 Subject: ASoC: Remove BROKEN from mpc5200 kconfig The regression was fixed by commit 3e5b50165fd0be080044586f43fcdd460ed27610, so no need to mark this driver as BROKEN. Signed-off-by: Takashi Iwai --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e7dd79a1d8cd..9fc908283371 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -21,7 +21,7 @@ config SND_SOC_MPC8610_HPCD config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" - depends on PPC_MPC52xx && PPC_BESTCOMM && BROKEN + depends on PPC_MPC52xx && PPC_BESTCOMM select SND_SOC_OF_SIMPLE select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.2 From 97a775c49c7e1b47b016a492463486a5b86da479 Mon Sep 17 00:00:00 2001 From: Jinyoung Park Date: Fri, 1 May 2009 12:54:31 +0100 Subject: ASoC: Fix errors in WM8990 The mis-typing exist in dapm controller definitions and dapm route definitions, so happen mis-matched error when snd_soc_dapm_add_routes(). Cc: stable@kernel.org Signed-off-by: Jinyoung Park Signed-off-by: Mark Brown Date: Mon, 11 May 2009 13:04:55 +0300 Subject: ASoC: soc-core: fix crash when removing not instantiated card If the card was not instantiated in snd_soc_instantiate_card, calling soc-remove will crash because some of codec, cpu_dai and card .remove methods are called twice. Fix this by returning from soc_remove immediately. Signed-off-by: Mike Rapoport Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f652d0d..1cd149b9ce69 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -954,6 +954,9 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; + if (!card->instantiated) + return 0; + run_delayed_work(&card->delayed_work); if (platform->remove) -- cgit v1.2.2 From 82075af6cb9b4918ab52a7100425b09fae6aafe3 Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 12:41:22 -0700 Subject: ASoC: davinci-pcm buildfixes This is a buildfix for the DaVinci PCM code, resyncing it with the version in the DaVinci tree. The notable change is using current EDMA interfaces, which recently merged to mainline. (The older interfaces never made it into mainline.) NOTE: open issue, the DMA should be to/from SRAM; see chip errata for more info. The artifacts are extremely easy to hear on DM355 hardware (not yet supported in mainline), but don't seem as audible on DM6446 hardwaare (which does have mainline support). Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-pcm.c | 71 ++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 7af3b5b3a53d..a05996588489 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -22,6 +22,7 @@ #include #include +#include #include "davinci-pcm.h" @@ -51,7 +52,7 @@ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ int master_lch; /* Master DMA channel */ - int slave_lch; /* Slave DMA channel */ + int slave_lch; /* linked parameter RAM reload slot */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } - davinci_set_dma_src_params(lch, src, INCR, W8BIT); - davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); - davinci_set_dma_src_index(lch, src_bidx, 0); - davinci_set_dma_dest_index(lch, dst_bidx, 0); - davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_src(lch, src, INCR, W8BIT); + edma_set_dest(lch, dst, INCR, W8BIT); + edma_set_src_index(lch, src_bidx, 0); + edma_set_dest_index(lch, dst_bidx, 0); + edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; @@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; - int tcc = TCC_ANY; + struct edmacc_param p_ram; int ret; if (!dma_data) @@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->params = dma_data; /* Request master DMA channel */ - ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, - &prtd->master_lch, &tcc, EVENTQ_0); - if (ret) + EVENTQ_0); + if (ret < 0) return ret; + prtd->master_lch = ret; - /* Request slave DMA channel */ - ret = davinci_request_dma(PARAM_ANY, "Link", - NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); - if (ret) { - davinci_free_dma(prtd->master_lch); + /* Request parameter RAM reload slot */ + ret = edma_alloc_slot(EDMA_SLOT_ANY); + if (ret < 0) { + edma_free_channel(prtd->master_lch); return ret; } - - /* Link slave DMA channel in loopback */ - davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + prtd->slave_lch = ret; + + /* Issue transfer completion IRQ when the channel completes a + * transfer, then always reload from the same slot (by a kind + * of loopback link). The completion IRQ handler will update + * the reload slot with a new buffer. + * + * REVISIT save p_ram here after setting up everything except + * the buffer and its length (ccnt) ... use it as a template + * so davinci_pcm_enqueue_dma() takes less time in IRQ. + */ + edma_read_slot(prtd->slave_lch, &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); + p_ram.link_bcntrld = prtd->slave_lch << 5; + edma_write_slot(prtd->slave_lch, &p_ram); return 0; } @@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_start_dma(prtd->master_lch); + edma_start(prtd->master_lch); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_stop_dma(prtd->master_lch); + edma_stop(prtd->master_lch); break; default: ret = -EINVAL; @@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct paramentry_descriptor temp; + struct edmacc_param temp; prtd->period = 0; davinci_pcm_enqueue_dma(substream); - /* Get slave channel dma params for master channel startup */ - davinci_get_dma_params(prtd->slave_lch, &temp); - davinci_set_dma_params(prtd->master_lch, &temp); + /* Copy self-linked parameter RAM entry into master channel */ + edma_read_slot(prtd->slave_lch, &temp); + edma_write_slot(prtd->master_lch, &temp); return 0; } @@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) spin_lock(&prtd->lock); - davinci_dma_getposition(prtd->master_lch, &src, &dst); + edma_get_position(prtd->master_lch, &src, &dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) count = src - runtime->dma_addr; else @@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + edma_unlink(prtd->slave_lch); - davinci_free_dma(prtd->slave_lch); - davinci_free_dma(prtd->master_lch); + edma_free_slot(prtd->slave_lch); + edma_free_channel(prtd->master_lch); kfree(prtd); -- cgit v1.2.2 From a62114cb90a351016121bca02e69d6a9e24afa0e Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 12:47:42 -0700 Subject: ASoC: DaVinci I2S updates This resyncs the DaVinci I2S code with the version in the DaVinci tree. The behavioral change uses updated clock interfaces which recently merged to mainline. Two other changes include adding a comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in order to support more boards than just the DM644x EVM. Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ffdb9439d3d8..b1ea52fc83c7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -24,6 +24,26 @@ #include "davinci-pcm.h" + +/* + * NOTE: terminology here is confusing. + * + * - This driver supports the "Audio Serial Port" (ASP), + * found on dm6446, dm355, and other DaVinci chips. + * + * - But it labels it a "Multi-channel Buffered Serial Port" + * (McBSP) as on older chips like the dm642 ... which was + * backward-compatible, possibly explaining that confusion. + * + * - OMAP chips have a controller called McBSP, which is + * incompatible with the DaVinci flavor of McBSP. + * + * - Newer DaVinci chips have a controller called McASP, + * incompatible with ASP and with either McBSP. + * + * In short: this uses ASP to implement I2S, not McBSP. + * And it won't be the only DaVinci implemention of I2S. + */ #define DAVINCI_MCBSP_DRR_REG 0x00 #define DAVINCI_MCBSP_DXR_REG 0x04 #define DAVINCI_MCBSP_SPCR_REG 0x08 @@ -421,7 +441,7 @@ static int davinci_i2s_probe(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[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, cpu_dai->private_data = dev; - dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -483,7 +503,7 @@ static void davinci_i2s_remove(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[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; -- cgit v1.2.2 From f492ec9f02908579353e31949855f86909a5af14 Mon Sep 17 00:00:00 2001 From: David Brownell Date: Thu, 14 May 2009 13:01:59 -0700 Subject: ASoC: DaVinci EVM board support buildfixes This is a build fix, resyncing the DaVinci EVM ASoC board code with the version in the DaVinci tree. That resync includes support for the DM355 EVM, although that board isn't yet in mainline. (NOTE: also includes a bugfix to the platform_add_resources call, recently sent by Chaithrika U S but not yet merged into the DaVinci tree.) Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 7 +++-- sound/soc/davinci/davinci-evm.c | 63 ++++++++++++++++++++++++++++++++++------- 2 files changed, 56 insertions(+), 14 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index bd7392c9657e..411a710be660 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + depends on SND_DAVINCI_SOC + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci EVM platform. + DaVinci DM6446 or DM355 EVM platforms. config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9b90b347007c..58fd1cbedd88 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,7 +20,11 @@ #include #include -#include +#include + +#include +#include +#include #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" @@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = { /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 0, + .i2c_bus = 1, .i2c_address = 0x1b, }; @@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ static struct resource evm_snd_resources[] = { { - .start = DAVINCI_MCBSP_BASE, - .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, .flags = IORESOURCE_MEM, }, }; static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DM644X_DMACH_MCBSP_TX, - .rx_dma_ch = DM644X_DMACH_MCBSP_RX, + .tx_dma_ch = DAVINCI_DMA_ASP0_TX, + .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +}; + +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm335evm_snd_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data dm335evm_snd_data = { + .tx_dma_ch = DAVINCI_DMA_ASP1_TX, + .rx_dma_ch = DAVINCI_DMA_ASP1_RX, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { + struct resource *resources; + unsigned num_resources; + struct evm_snd_platform_data *data; + int index; int ret; - evm_snd_device = platform_device_alloc("soc-audio", 0); + if (machine_is_davinci_evm()) { + davinci_cfg_reg(DM644X_MCBSP); + + resources = evm_snd_resources; + num_resources = ARRAY_SIZE(evm_snd_resources); + data = &evm_snd_data; + index = 0; + } else if (machine_is_davinci_dm355_evm()) { + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + resources = dm335evm_snd_resources; + num_resources = ARRAY_SIZE(dm335evm_snd_resources); + data = &dm335evm_snd_data; + index = 1; + } else + return -EINVAL; + + evm_snd_device = platform_device_alloc("soc-audio", index); if (!evm_snd_device) return -ENOMEM; platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, &evm_snd_data, - sizeof(evm_snd_data)); + platform_device_add_data(evm_snd_device, data, sizeof(*data)); - ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, - ARRAY_SIZE(evm_snd_resources)); + ret = platform_device_add_resources(evm_snd_device, resources, + num_resources); if (ret) { platform_device_put(evm_snd_device); return ret; -- cgit v1.2.2