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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[alsa-devel] [PATCH] ALSA: ASoC: fix NULL pointer dereference in soc_suspend()
by Daniel Mack 04 Jun '09

04 Jun '09

In case the initalization of an soc_device failed, there is no codec associated with it. soc_suspend() will still dereference the pointer and cause an Ooops when entering the sleep mode. This happens on our board with a multi-target kernel image when booted on a machine without audio circuits. This patch makes the code bail out very early in this special case. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1cd149b..4545a15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -632,6 +632,12 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_codec *codec = card->codec; int i; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!codec) + return 0; + /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ -- 1.6.3.1

2 1

[alsa-devel] [PATCH] hda_intel: fix build error when !PM
by Alexander Beregalov 04 Jun '09

04 Jun '09

Fix this build error when CONFIG_PM is not set: ound/pci/hda/hda_intel.c: In function 'azx_bus_reset': sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all' sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend' sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume' Signed-off-by: Alexander Beregalov <a.beregalov(a)gmail.com> --- sound/pci/hda/hda_intel.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f44645..4e9ea70 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1260,17 +1260,20 @@ static void azx_stop_chip(struct azx *chip); static void azx_bus_reset(struct hda_bus *bus) { struct azx *chip = bus->private_data; - int i; bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip); +#ifdef CONFIG_PM if (chip->initialized) { + int i; + for (i = 0; i < AZX_MAX_PCMS; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus); snd_hda_resume(chip->bus); } +#endif bus->in_reset = 0; } -- 1.6.3.1

2 1

[alsa-devel] Wonder if VT1708A has Mic Boost or not
by Konstantin Kletschke 04 Jun '09

04 Jun '09

Hi Folks! I have a mini-PC here which should record Voice over Microphone. The Level is not really loud enough and the alsa-driver-1.0.19 doesn't reveal a mic boost setting in its mixer. This of course may be of the driver not yet being adapted to the special motherboard AFAIK. What I wonder is, is the Mic Boost setting/function integrated in this chip itself or does hardware vendor need to implement external opamps for this? If it is integrated into the chip itself I could try to modifiy the code to make the setting public. Is there a source for a datasheet anywhere for the VT1708A? King regards, Konsti -- GPG KeyID EF62FCEF Fingerprint: 13C9 B16B 9844 EC15 CC2E A080 1E69 3FDA EF62 FCEF

2 2

[alsa-devel] [PATCH v3 5/6] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Chaithrika U S 04 Jun '09

04 Jun '09

Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). The codec on the DM646x EVM is AIC32 which is connected to Serializer0 and Serializer1 of McASP0. The McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. 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/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-i2s-mcasp.c | 908 +++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-i2s-mcasp.h | 58 +++ sound/soc/davinci/davinci-pcm.h | 17 +- 5 files changed, 991 insertions(+), 8 deletions(-) create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.c create mode 100644 sound/soc/davinci/davinci-i2s-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 411a710..7d10a8c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_I2S_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" depends on SND_DAVINCI_SOC @@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_I2S_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..152d59c 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-i2s-mcasp-objs:= davinci-i2s-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S_MCASP) += snd-soc-davinci-i2s-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-i2s-mcasp.c b/sound/soc/davinci/davinci-i2s-mcasp.c new file mode 100644 index 0000000..6ea75eb --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.c @@ -0,0 +1,908 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * 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 + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left (even TDM Slot) Channel Status Register File*/ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right (odd TDM slot) Channel Status RegisterFile*/ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left (even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right (odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_i2s_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct snd_pcm_substream *substream, + int channel_size) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -1; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + 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 */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + 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); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_i2s_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(substream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(substream); + else + davinci_hw_param(substream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(substream, word_length); + + return 0; +} + +static int davinci_i2s_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(substream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_mcasp_trigger, + .hw_params = davinci_i2s_mcasp_hw_params, + .set_fmt = davinci_i2s_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_iis_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_i2s_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_i2s_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_iis_mcasp_dai); + +static int davinci_i2s_mcasp_probe(struct platform_device *pdev) +{ + struct evm_snd_platform_data *parray = pdev->dev.platform_data; + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct evm_snd_platform_data *pdata; + int tmp, link_cnt, num_links = 2; + struct davinci_audio_dev *dev; + int count = 0; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev) * num_links, + GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * + (num_links << 1), GFP_KERNEL); + + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + for (link_cnt = 0; link_cnt < num_links; link_cnt++) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, link_cnt); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = &parray[link_cnt]; + dev[link_cnt].clk = clk_get(&pdev->dev, pdata->clk_name); + + if (IS_ERR(dev[link_cnt].clk)) { + ret = -ENODEV; + goto err_release_region; + } + + clk_enable(dev[link_cnt].clk); + + dev[link_cnt].base = (void __iomem *)IO_ADDRESS(mem->start); + dev[link_cnt].op_mode = pdata->op_mode; + dev[link_cnt].tdm_slots = pdata->tdm_slots; + dev[link_cnt].num_serializer = pdata->num_serializer; + dev[link_cnt].serial_dir = pdata->serial_dir; + dev[link_cnt].codec_fmt = pdata->codec_fmt; + + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_PLAYBACK] = + &dma_data[count]; + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, + 2 * link_cnt + 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev[link_cnt].base)); + dev[link_cnt].dma_params[SNDRV_PCM_STREAM_CAPTURE] = + &dma_data[count]; + + res = platform_get_resource(pdev, IORESOURCE_DMA, + 2 * link_cnt + 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + davinci_iis_mcasp_dai[link_cnt].private_data = &dev[link_cnt]; + } + ret = snd_soc_register_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + for (tmp = link_cnt; tmp >= 0; tmp--) { + mem = platform_get_resource(pdev, IORESOURCE_MEM, tmp); + release_mem_region(mem->start, + (mem->end - mem->start) + 1); + } +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_i2s_mcasp_remove(struct platform_device *pdev) +{ + int i, num_links = 2; + struct davinci_audio_dev *dev; + struct resource *mem; + struct davinci_pcm_dma_params *dma_data; + + snd_soc_unregister_dais(davinci_iis_mcasp_dai, + ARRAY_SIZE(davinci_iis_mcasp_dai)); + + for (i = 0; i < num_links; i++) { + dev = davinci_iis_mcasp_dai[i].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, i); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + } + + dev = davinci_iis_mcasp_dai[0].private_data; + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_i2s_mcasp_probe, + .remove = davinci_i2s_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_i2s_mcasp_init(void) +{ + return platform_driver_register(&davinci_mcasp_driver); +} +module_init(davinci_i2s_mcasp_init); + +static void __exit davinci_i2s_mcasp_exit(void) +{ + platform_driver_unregister(&davinci_mcasp_driver); +} +module_exit(davinci_i2s_mcasp_exit); + + + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McASP) SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-i2s-mcasp.h b/sound/soc/davinci/davinci-i2s-mcasp.h new file mode 100644 index 0000000..f881e54 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s-mcasp.h @@ -0,0 +1,58 @@ +/* + * ALSA SoC I2S (McASP) Audio Layer for TI DAVINCI processor + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen, <schen@.mvista.com> + * + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2008 Texas Instruments, India + * + * 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 + * published by the Free Software Foundation. + */ + + +#ifndef DAVINCI_I2S_MCASP_H +#define DAVINCI_I2S_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_iis_mcasp_dai[]; +extern struct snd_soc_dai davinci_dit_mcasp_dai; + + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + + + +#endif /* DAVINCI_I2S_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..3cbe281 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,18 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

3 5

[alsa-devel] [PATCH] ALSA: Add missing __devexit_p() markers
by Jean Delvare 04 Jun '09

04 Jun '09

3 ISA sound drivers lack their __devexit_p() markers, which would cause build failures when the kernel is built without hotplug support. Signed-off-by: Jean Delvare <khali(a)linux-fr.org> Cc: Kyle McMartin <kyle(a)mcmartin.ca> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/es1688/es1688.c | 2 +- sound/isa/gus/gusextreme.c | 2 +- sound/parisc/harmony.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) --- linux-2.6.30-rc8.orig/sound/isa/es1688/es1688.c 2009-04-08 08:54:02.000000000 +0200 +++ linux-2.6.30-rc8/sound/isa/es1688/es1688.c 2009-06-04 10:37:06.000000000 +0200 @@ -193,7 +193,7 @@ static int __devexit snd_es1688_remove(s static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, .probe = snd_es1688_probe, - .remove = snd_es1688_remove, + .remove = __devexit_p(snd_es1688_remove), #if 0 /* FIXME */ .suspend = snd_es1688_suspend, .resume = snd_es1688_resume, --- linux-2.6.30-rc8.orig/sound/isa/gus/gusextreme.c 2009-04-08 08:54:02.000000000 +0200 +++ linux-2.6.30-rc8/sound/isa/gus/gusextreme.c 2009-06-04 10:37:13.000000000 +0200 @@ -348,7 +348,7 @@ static int __devexit snd_gusextreme_remo static struct isa_driver snd_gusextreme_driver = { .match = snd_gusextreme_match, .probe = snd_gusextreme_probe, - .remove = snd_gusextreme_remove, + .remove = __devexit_p(snd_gusextreme_remove), #if 0 /* FIXME */ .suspend = snd_gusextreme_suspend, .resume = snd_gusextreme_resume, --- linux-2.6.30-rc8.orig/sound/parisc/harmony.c 2009-04-08 08:54:03.000000000 +0200 +++ linux-2.6.30-rc8/sound/parisc/harmony.c 2009-06-04 10:37:25.000000000 +0200 @@ -1020,7 +1020,7 @@ static struct parisc_driver snd_harmony_ .name = "harmony", .id_table = snd_harmony_devtable, .probe = snd_harmony_probe, - .remove = snd_harmony_remove, + .remove = __devexit_p(snd_harmony_remove), }; static int __init -- Jean Delvare

2 1

[alsa-devel] snd-ctxfi
by The Source 04 Jun '09

04 Jun '09

Hello. I tested the snd-ctxfi driver and found no problems (except no support for 5.1 and external console, but original ctxfi does not support them either). A minor problem: kernel oops reporter periodically pops up with message about using a timer in a wrong place. I'll post here this message when I get it next time.

3 7

[alsa-devel] [HELP] snd-aloop with dmix?
by Santo Chow 04 Jun '09

04 Jun '09

HI all, I'm trying to use snd-aloop as my default soundard. I have the .asoundrc that look like this. pcm.sndcard{ type hw card 1 #snd-aloop soundcard device 0 } pcm.dmixer { type dmix ipc_key 1025 ipc_perm 0660 slave { pcm sndcard period_time 0 period_size 881 buffer_size 3524 rate 44100 } bindings { 0 0 1 1 } } pcm.dsnooper { type dsnoop ipc_key 1026 ipc_perm 0660 slave { pcm sndcard period_time 0 period_size 881 buffer_size 3524 rate 44100 } bindings { 0 0 1 1 } } pcm.duplexasym { type asym playback.pcm "dmixer" # playback capture.pcm "dsnooper" # capture } pcm.!default { type plug slave { pcm duplexasym } } ctl.duplex { type hw card 1 } Right now my problem is, i can only get 1 sound from an application at a time. I can't run several applications at the same time and recorded all the sounds. This is what I did : My default sound card is indexed at 0, so the snd-aloop will be at index 1. To test this, I first run an MP3 player. Then I run this in terminal: arecord -D hw:1,1 -f cd | aplay -D hw:0 -f cd The sound come out good. Then when I open a browser, go to youtube and play a song over there, the sound won't come out. I have to first close the browser and the currently playing MP3 player, then re-open a browser, go to youtube, and play a song, the sound will come out. So, my question is.. is there something missing in my .asoundrc file? can the snd-aloop support multiple playbacks and captures at the same time? is there a way for me to achieve this?

1 0

[alsa-devel] Requesting opinions and experiences concerning laptops and external microphones
by komputes 04 Jun '09

04 Jun '09

I am trying to gather opinions and experiences from the ALSA developer community to understand what they expect to happen when they plug in an analog microphone into their laptops (which already has a built in mic). I am looking for more of an official ALSA document which states how capture should change when an analogue microphone is plugged in; should it takeover from the internal microphone or not? Is there specification, standard, or "correct" behavior which is applied to this process? If I am not mistaking, the computer can detect a signal/message when the jack goes in, right? The expected behavior when plugging in headphones is that the headphones are detected, the speakers are muted and the audio is channeled to the headphones. Is this not applicable to external microphone and the internal microphone as well? -komputes

2 1

Re: [alsa-devel] ALSA patch Nvidia 8 channel hdmi audio
by Marek Vasut 04 Jun '09

04 Jun '09

On Wednesday 03 of June 2009 00:30:04 Alan Alan wrote: > Thank's for the info. > How can i contact him ? He and alsa-devel mailing list are both in CC already, please preserve them there or fix your mailer. > > > 2009/6/3 Marek Vasut <marek.vasut(a)gmail.com> > > > I'm not the author of the patch, CCing alsa-devel and the original > > author. > > > > On Tuesday 02 of June 2009 23:58:26 Alan Alan wrote: > > > Hi Marek ! > > > > > > I just read your emailing with nvidia devs about alsa supporting 8 > > > > channel > > > > > lpcm over hdmi. > > > > > > I checked it out an i have to tell it is working awsome, except for a > > > few bugs: > > > > > > The channels are wrong mapped. Running a speakertest -Dhdmi -c6 gives > > > the following result (first where it should sound) > > > FL - FL > > > C - RL > > > FR - FR > > > RR - LFE > > > RL - C > > > > > > The other thing i realized that playback is limited to 16 bits 48 khz. > > > I think the minimum which is supported by nvidia hardware is 24bits 96 > > > (or 192) khz. > > > Why not ? > > > > > > Could you contact the devs as i don't know how to reach them. > > > > > > ps.: Mybe my build methot was not 100% ok ? I used 1.020 Alsa with the > > > oatched nvidia.c file. Anyway the amplifier shows multichannel LPCM and > > > that's great news.

6 17

[alsa-devel] [PATCH] ALSA: Fix double locking of card list in snd_card_register()
by Mark Brown 04 Jun '09

04 Jun '09

The introduction of snd_card_set_id() added a lock on the card list to the old choose_default_id() function when using it to implement the new API call. This lock is needed to allow us to walk the list and check to see if our new name is a duplicate. Unfortunately this causes a lockup when called from snd_card_register() (in cases where no ID is supplied for the card) since the card list is already locked there. Fix this fairly hideously by factoring out the implementation and using a flag to indicate if the lock should be held. A better fix would probably be to refactor snd_card_register() to move the _set_id() outside the locking region but I can't immediately see anything I can convince myself is safe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This patch isn't terribly nice but I figured it was a more useful report than just the deadlock. sound/core/init.c | 31 +++++++++++++++++++------------ 1 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 6557dd8..a578d05 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -476,15 +476,8 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -/** - * snd_card_set_id - set card identification name - * @card: soundcard structure - * @nid: new identification string - * - * This function sets the card identification and checks for name - * collisions. - */ -void snd_card_set_id(struct snd_card *card, const char *nid) +static void snd_card_set_id_internal(struct snd_card *card, const char *nid, + int do_locking) { int i, len, idx_flag = 0, loops = SNDRV_CARDS; const char *spos, *src; @@ -529,14 +522,16 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } if (!snd_info_check_reserved_words(id)) goto __change; - mutex_lock(&snd_card_mutex); + if (do_locking) + mutex_lock(&snd_card_mutex); for (i = 0; i < snd_ecards_limit; i++) { if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) { mutex_unlock(&snd_card_mutex); goto __change; } } - mutex_unlock(&snd_card_mutex); + if (do_locking) + mutex_unlock(&snd_card_mutex); break; __change: @@ -561,6 +556,18 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } } +/** + * snd_card_set_id - set card identification name + * @card: soundcard structure + * @nid: new identification string + * + * This function sets the card identification and checks for name + * collisions. + */ +void snd_card_set_id(struct snd_card *card, const char *nid) +{ + snd_card_set_id_internal(card, nid, 1); +} EXPORT_SYMBOL(snd_card_set_id); #ifndef CONFIG_SYSFS_DEPRECATED @@ -657,7 +664,7 @@ int snd_card_register(struct snd_card *card) return 0; } if (card->id[0] == '\0') - snd_card_set_id(card, NULL); + snd_card_set_id_internal(card, NULL, 0); snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- 1.6.3.1

3 3

[alsa-devel] [PATCH] More Aspire 8930G fixes
by Hector Martin 04 Jun '09

04 Jun '09

Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c.orig 2009-06-03 23:17:55.000000000 +0200 +++ alsa-driver-1.0.20.10.g955ee.233.gf65b3/sound/pci/hda/patch_realtek.c 2009-06-03 23:45:40.000000000 +0200 @@ -1469,10 +1469,10 @@ }; /* - * ALC888 Acer Aspire 8930G model + * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -1492,7 +1492,7 @@ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect HP out to Front */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Enable all DACs */ @@ -1507,6 +1507,17 @@ /* Enable amplifiers */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, +/* DMIC fix + * This laptop has a stereo digital microphone. The mics are only 1cm apart + * which makes the stereo useless. However, either the mic or the ALC889 + * makes the signal become a difference/sum signal instead of standard + * stereo, which is annoying. So instead we flip this bit which makes the + * codec replicate the sum signal to both channels, turning it into a + * normal mono mic. + */ +/* DMIC_CONTROL? Init value = 0x0001 */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0003}, { } }; @@ -1531,6 +1542,38 @@ } }; +static struct hda_input_mux alc889_capture_sources[3] = { + /* Digital mic only available on first "ADC" */ + { + .num_items = 5, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Front Mic", 0xb }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + }, + { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + { "Input Mix", 0xa }, + }, + } +}; + static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1562,7 +1605,7 @@ alc_automute_amp(codec); } -static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9081,22 +9124,22 @@ .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc888_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), - .adc_nids = alc883_adc_nids_rev, - .capsrc_nids = alc883_capsrc_nids_rev, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .const_channel_count = 6, .num_mux_defs = - ARRAY_SIZE(alc888_2_capture_sources), - .input_mux = alc888_2_capture_sources, + ARRAY_SIZE(alc889_capture_sources), + .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_8930g_init_hook, + .init_hook = alc889_acer_aspire_8930g_init_hook, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer,

2 1

[alsa-devel] [PATCH] burgundy: timeout message is off by one.
by Roel Kluin 04 Jun '09

04 Jun '09

Timeout message is off by one. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 45a7629..d2c0a4e 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -46,12 +46,12 @@ snd_pmac_burgundy_extend_wait(struct snd_pmac *chip) timeout = 50; while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n"); timeout = 50; while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n"); }

2 1

[alsa-devel] [PATCH - quirk for USB Aureon cards 1/1] Add quirk to provide proper naming of the Terratec Aureon 5.1 MkII USB card
by Andrea Borgia 03 Jun '09

03 Jun '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 5d955aa..d84d6f3 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1951,6 +1951,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0028), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "TerraTec", + .product_name = "Aureon 5.1 MkII", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", -- 1.6.0.4

2 8

[alsa-devel] Sequencer timestamps for soft synth app
by Glyn Adgie 03 Jun '09

03 Jun '09

I have written a soft synth, which uses ALSA to get MIDI events, and Jack to do audio output. MIDI events are fetched and processed at the start of the Jack audio callback, then the Jack output buffers are filled by the synth engine. The problem I have is that, if the Jack buffer size is too large, the time resolution for MIDI events is too coarse, and this has a severe effect on rhythms. I currently get around this by setting up Jack for a buffer size of 64 frames, and a sample rate of 48 kHz, giving a timing resolution of 1.33 msec, which appears to be adequate. However, I might not always be able to get the desired buffer size. If the Jack buffer is some multiple of 64 frames, the synth renders multiple 64 frame chunks until the buffer is filled. This takes care of audio quality. However, MIDI timing resolution suffers. I am looking at using the event's timestamp, to decide when to process an event. For example, I might have a Jack buffer of 256 frames, so I want to work out whether to process an event right away, or after one 64 frame chunk, and so on. It would appear that, in order to make this work, I need an initial timestamp, which I get at the start of the audio callback. I then compare each event's timestamp, to determine where it should be processed. I cannot find an ALSA function for getting a timestamp. Perhaps I am missing something, or trying to do the job in the wrong way. Regards Glyn

2 1

[alsa-devel] Xonar DX spdif advice request
by Dmitri Seletski 03 Jun '09

03 Jun '09

Hello guys. I have perfectly working asus xonar card. it works fine with ordinary outputs, but I would like to know, did anyone try those cards with spdif? Is it worth buying something spdif compatible, if you intend to use this card? Thank you in advance. Dmitri

1 0