June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoc: codec tlv320aic23 fix bogus divide by 0 message
by Troy Kisky 06 Jun '09

06 Jun '09

Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/codecs/tlv320aic23.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 9fcbb9c..0b8dcb5 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = { * Every divisor is a factor of 11*12 */ #define SR_MULT (11*12) -#define A(x) (x) ? (SR_MULT/x) : 0 +#define A(x) (SR_MULT/x) static const unsigned char sr_adc_mult_table[] = { - A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), - A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) + A(2), A(2), A(12), A(12), 0, 0, A(3), A(1), + A(2), A(2), A(11), A(11), 0, 0, 0, A(1) }; static const unsigned char sr_dac_mult_table[] = { - A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), - A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) + A(2), A(12), A(2), A(12), 0, 0, A(3), A(1), + A(2), A(11), A(2), A(11), 0, 0, 0, A(1) }; static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, -- 1.5.4.3

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Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Paul Shen 06 Jun '09

06 Jun '09

Hi Daniel, This is Paul Shen , I will substitute boshen9(a)gmail.com for bshen9(a)marvell.com to answer open source mails. > From: Daniel Mack <daniel(a)caiaq.de> > Date: Wed, Jun 3, 2009 at 9:18 PM > Subject: Re: [alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support > To: Eric Miao <eric.y.miao(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org, linux-arm-kernel <linux-arm-kernel(a)lists.arm.linux.org.uk>, Mark Brown <broonie(a)sirena.org.uk> > > > On Wed, Jun 03, 2009 at 08:33:42PM +0800, Eric Miao wrote: > > Make the pxa I2S configuration generic, add support for Left_J, add > > support for variable frame width like 32fs, 48fs, 64fs and 96fs > > > > Signed-off-by: Paul Shen <bshen9(a)marvell.com> > > Signed-off-by: Eric Miao <eric.miao(a)marvell.com> > > Cc: Daniel Mack <daniel(a)caiaq.de> > > --- > > arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- > > sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- > > sound/soc/pxa/pxa-ssp.h | 9 ++++ > > 3 files changed, 47 insertions(+), 38 deletions(-) > > Ok, I tried that code on my board and this is what I had to change > there: > > The tdm time slot configuration needs to be set again in my board support > code just like in your example: snd_soc_set_tdm_slot(cpu_dai, 3, 2). And > the PXA_SSP_DIV_SCR value needed to be doubled from 4 to 8. > I tested with below codes to set the cpu_dai : format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; snd_soc_dai_set_fmt(cpu_dai, format); snd_soc_dai_set_tdm_slot(cpu_dai, 3, 2); snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 0); snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); It is no need to "snd_soc_dai_set_pll(cpu_dai, 0, 0, clk)" , for your case ,no ssp-on-chip pll configuration is needed. On my littleton platform,it is ok, and I did not modify the PXA_SSP_DIV_SCR from 4 to 8. I dumped the ssp register: SSCR0 0xa10003ff SSCR1 0x00e01d80 SSTO 0x00000000 SSPSP 0x31a00084 SSSR 0x0000f0fc SSACD 0x00000000 SSACDD 0x00000000 How about your registers ? > With that changes, LRCLK is 44100Hz when configured to 44100Hz. But the > bitclk is not 64fs anymore but 32fs only (1.41Mhz). Is there any > implementation details I miss? What does your codec clock config look > like? > Do you mean it only down to 32fs when LRCLK is 44100HZ ? And when LRCLK is 48000HZ, all is OK ? > Another small thing: > > CC sound/soc/pxa/pxa-ssp.o > sound/soc/pxa/pxa-ssp.c:186: warning: 'ssp_get_scr' defined but not used > > Daniel > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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[alsa-devel] [PATCH] Palm interrupt driven touchscreen driver
by Marek Vasut 05 Jun '09

05 Jun '09

Hi, this patch adds interrupt driven touchscreen driver. It's an addition to mainstone-wm97xx, but does some reorganization of the code there. Mark, it's the same patch we discussed on IRC, so it should be OK. Please consider applying, it's against Eric's tree. Thanks

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[alsa-devel] [PATCH v4 4/4] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 05 Jun '09

05 Jun '09

This patch does the following: (1) Add support for the DM646x machine (2) Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-evm.c | 102 ++++++++++++++++++-------------------- 1 files changed, 48 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..f3bb6f6 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -27,9 +27,10 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +44,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +145,24 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +171,26 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,60 +198,30 @@ 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_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .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, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; 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; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; 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; + evm_snd_dev_data = &evm_snd_devdata; 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; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +229,8 @@ static int __init evm_init(void) 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, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

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[alsa-devel] M-Audio Transit: Running asynchronous USB?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. I would like to get a confirmation on below. One of my USB soundcards is a M-Audio Transit. cat /proc/asound/card0/stream0 lists below: M-Audio Transit USB at usb-0000:00:1d.2-1.3, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 3 URBs = 2 [ 1 1 ] Packet Size = 196 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous) Interface 1 Altset 2 Format: 0x20 (24 bits) Channels: 2 Endpoint: 3 OUT (NONE) Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: 0x2 (16 bits) Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Do I interprete it right that the Transit runs in asynchronous USB mode up to 48khz and 16bit? Does above also mean that <48/24 is not supported? THX Klaus

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[alsa-devel] How to send control codes to a USB device?
by Klaus Schulz 05 Jun '09

05 Jun '09

Hi there. Is there a way to sent control codes to a USB device, without using a specific driver . I would like to try to set latency and output modes of my M-Audio Transit by sending a control code as additional part of the UDEV madfuload firmware initialzation. How can I do something like that? If this works in general I just need to figure out the required control codes. THX for your support Klaus

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[alsa-devel] [PATCH] Fix possibly incorrect offset in wm97xx-core
by Marek Vasut 05 Jun '09

05 Jun '09

Hi, I just hit a problem with the accelerated (interrupt driven) ts driver. It stop responding after a while (sometimes right after you release a stylus off the touchscreen, it doesn't generate any more interrupts). I made the following patch that fixes the problem for me (and if I read the 9712 and 9713 manuals correctly, the patch should be OK) Thanks for reply. diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index 69af838..4f13a7f 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -307,11 +307,7 @@ static void wm97xx_pen_irq_worker(struct work_struct *work) WM97XX_GPIO_13); } - if (wm->id == WM9712_ID2) - wm97xx_reg_write(wm, AC97_GPIO_STATUS, (status & - ~WM97XX_GPIO_13) << 1); - else - wm97xx_reg_write(wm, AC97_GPIO_STATUS, status & + wm97xx_reg_write(wm, AC97_GPIO_STATUS, status & ~WM97XX_GPIO_13); mutex_unlock(&wm->codec_mutex); }

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[alsa-devel] ASoC update for 2.6.31
by Mark Brown 05 Jun '09

05 Jun '09

The following changes since commit d08664fdb50795b29cf70b0269ea02f7248e76c3: Takashi Iwai (1): ASoC: Fix build error in twl4030.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Daniel Mack (1): ASoC: fix NULL pointer dereference in soc_suspend() sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

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[alsa-devel] ALSA 1.0.20 Speex PCM Plugin
by Robert Krakora 04 Jun '09

04 Jun '09

Hello, Has anyone successfully employed the ALSA 1.0.20 Speex PCM Plugin? I followed the "speexdsp.txt" document under the 'doc' directory but the result was the following error: [root@vizioroom105 ~]# arecord -Dplug:mic poopy.wav Recording WAVE 'poopy.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable arecord: set_params:957: Broken configuration for this PCM: no configurations available Here is my asound.conf: # MessageNet Systems ALSA sound configuration file # Allowing multiple playback and capture pcm.!default { type plug slave.pcm "duplex" } pcm.duplex { type asym playback.pcm "dmixer" capture.pcm "analog_input" } pcm.dmixer { type dmix ipc_key 1024 ipc_perm 0666 slave { pcm "hw:2,3" period_time 0 period_size 1024 buffer_size 8192 } bindings { 0 0 1 1 } } pcm.mic { type speex slave.pcm { type hw card 1 } agc 1 agc_level 8000 } pcm.analog_input { type dsnoop ipc_key 2048 ipc_perm 0666 slave{ pcm "hw:1,2" } } pcm.dsp0 { type plug slave.pcm "dmixer" } ctl.mixer0 { type hw card 0 } Here are the results of alsa-info.sh: http://www.alsa-project.org/db/?f=358e4d5d828dda3c44220382c8b480fd6019ab2a Thanks in advance. Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

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[alsa-devel] [PATCH 3/4] ASoC: pxa add Marvell Littleton platform audio support
by Eric Miao 04 Jun '09

04 Jun '09

Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/littleton.c | 228 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/littleton.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ad8a10f..a4ad5ba 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701 help Say Y if you want to add support for SoC audio on the MIO A701. + +config SND_PXA2XX_SOC_LITTLETON + tristate "SoC Audio support for Marvell littleton" + depends on SND_PXA2XX_SOC && MACH_LITTLETON + select SND_PXA_SOC_SSP + select SND_SOC_DA9034 + help + Say Y if you want to add support for SoC audio on the + Marvell Littleton. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 4b90c3c..94c719d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ 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 +snd-soc-littleton-objs := littleton.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -34,4 +35,5 @@ 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-$(CONFIG_SND_PXA2XX_SOC_LITTLETON) += snd-soc-littleton.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/littleton.c b/sound/soc/pxa/littleton.c new file mode 100644 index 0000000..911a5ef --- /dev/null +++ b/sound/soc/pxa/littleton.c @@ -0,0 +1,228 @@ +/* + * littleton.c -- SoC audio for Littleton + * + * Copyright (C) 2009 Marvell International Ltd. + * + * Author: Paul Shen <bshen9(a)marvell.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/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> +#include <mach/ssp.h> +#include <mach/regs-ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +static int hifi_mode; + +extern struct snd_soc_dai da9034_dais[2]; +extern struct snd_soc_codec_device soc_codec_dev_da9034; + +static const char *hifi_mode_select[] = { + "HIFI 64FS LEFT_J, SSP CLK Master FRM Master", + "HIFI 64FS I2S , SSP CLK Master FRM Master", + "HIFI 32FS LEFT_J, SSP CLK Master FRM Master", + "HIFI 32FS I2S , SSP CLK Master FRM Master", +}; + +static const struct soc_enum littleton_enum[] = { + SOC_ENUM_SINGLE_EXT(4, hifi_mode_select), +}; + +static int littleton_get_hifi_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hifi_mode; + return 0; +} + +static int littleton_set_hifi_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + hifi_mode = ucontrol->value.integer.value[0]; + return 1; +} + +static const struct snd_kcontrol_new littleton_da9034_controls[] = { + SOC_ENUM_EXT("Hifi Mode Select", littleton_enum[0], + littleton_get_hifi_mode, littleton_set_hifi_mode), +}; + +static int littleton_da9034_init(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, littleton_da9034_controls, + ARRAY_SIZE(littleton_da9034_controls)); + return snd_soc_dapm_sync(codec); +} + +struct ssp_clk_div { + unsigned int rate; + unsigned int pll; + unsigned int scdb; + unsigned int acds; +}; + +static struct ssp_clk_div littleton_hifi_clk_div_64fs[] = { + { 48000, 12288000, PXA_SSP_CLK_SCDB_4, 0 }, + { 44100, 11289600, PXA_SSP_CLK_SCDB_4, 0 }, + { 32000, 4096000, PXA_SSP_CLK_SCDB_1, 1 }, + { 22050, 11289600, PXA_SSP_CLK_SCDB_8, 0 }, + { 16000, 4096000, PXA_SSP_CLK_SCDB_4, 0 }, + { 11025, 11289600, PXA_SSP_CLK_SCDB_4, 2 }, + { 8000, 4096000, PXA_SSP_CLK_SCDB_4, 1 }, + {}, +}; + +static struct ssp_clk_div littleton_hifi_clk_div_32fs[] = { + { 48000, 12288000, PXA_SSP_CLK_SCDB_4, 1 }, + { 44100, 11289600, PXA_SSP_CLK_SCDB_4, 1 }, + { 32000, 4096000, PXA_SSP_CLK_SCDB_1, 2 }, + { 22050, 11289600, PXA_SSP_CLK_SCDB_8, 1 }, + { 16000, 4096000, PXA_SSP_CLK_SCDB_4, 1 }, + { 11025, 11289600, PXA_SSP_CLK_SCDB_8, 2 }, + { 8000, 4096000, PXA_SSP_CLK_SCDB_8, 1 }, + {}, +}; + +static int littleton_hifi_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; + struct ssp_clk_div *clkdiv; + struct clk *clk_pout; + int format; + + clk_pout = clk_get(NULL, "CLK_POUT"); + if (IS_ERR(clk_pout)) { + pr_err("failed to get POUT clock\n"); + return PTR_ERR(clk_pout); + } + + clk_enable(clk_pout); + + switch (hifi_mode) { + case 0: + format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; + clkdiv = littleton_hifi_clk_div_64fs; + break; + case 1: + format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_64FS; + clkdiv = littleton_hifi_clk_div_64fs; + break; + case 2: + format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_32FS; + clkdiv = littleton_hifi_clk_div_32fs; + break; + case 3: + format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | PXA_SSP_FRM_32FS; + clkdiv = littleton_hifi_clk_div_32fs; + break; + default: + return -EINVAL; + } + + snd_soc_dai_set_fmt(codec_dai, format); + snd_soc_dai_set_fmt(cpu_dai, format); + snd_soc_dai_set_tdm_slot(cpu_dai, 3, 2); + + for (; clkdiv->rate; clkdiv++) { + if (clkdiv->rate == params_rate(params)) + break; + } + + if (!clkdiv->rate) + return -EINVAL; + + snd_soc_dai_set_pll(cpu_dai, 0, 0, clkdiv->pll); + snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_SCDB, clkdiv->scdb); + snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, clkdiv->acds); + return 0; +} + +static struct snd_soc_ops littleton_hifi_ops = { + .hw_params = littleton_hifi_hw_params, +}; + +static struct snd_soc_dai_link littleton_dai[] = { + /* HiFi */ + { + .name = "DA9034 HiFi", + .stream_name = "DA9034 HiFi", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .codec_dai = &da9034_dais[0], + .init = littleton_da9034_init, + .ops = &littleton_hifi_ops, + }, +}; + +static struct snd_soc_card littleton = { + .name = "Littleton", + .platform = &pxa2xx_soc_platform, + .dai_link = littleton_dai, + .num_links = ARRAY_SIZE(littleton_dai), +}; + +static struct snd_soc_device littleton_snd_devdata = { + .card = &littleton, + .codec_dev = &soc_codec_dev_da9034, +}; + +static struct platform_device *littleton_snd_device; + +static int __init littleton_audio_init(void) +{ + int ret; + + if (!machine_is_littleton()) { + pr_err("Only Littleton hardware supported by ASoC driver\n"); + return -ENODEV; + } + + littleton_snd_device = platform_device_alloc("soc-audio", -1); + if (!littleton_snd_device) + return -ENOMEM; + + platform_set_drvdata(littleton_snd_device, &littleton_snd_devdata); + littleton_snd_devdata.dev = &littleton_snd_device->dev; + + ret = platform_device_add(littleton_snd_device); + if (ret != 0) + platform_device_put(littleton_snd_device); + + return ret; +} + +static void __exit littleton_audio_exit(void) +{ + platform_device_unregister(littleton_snd_device); +} + +module_init(littleton_audio_init); +module_exit(littleton_audio_exit); + +MODULE_AUTHOR("Paul Shen <bshen9(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC Da9034 Littleton"); +MODULE_LICENSE("GPL"); -- 1.6.0.4

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