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

[alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 25 May '09

25 May '09

Hi all, I'm new to ALSA ASoC and ALSA. I'd like get some advice where I can get more information about writing an ASoC driver. I've browsed the internet couples of days and I can only find links on ALSA driver api's and writing an ALSA driver for PCI devices. It seems there is not much information on how to writing an ALSA ASoC driver. I'm stuck on the following questions: 1. describing those important structures, e.g. struct soc_enum, struct snd_kcontrol_new, struct snd_soc_dapm_widget; 2. what is/are the relationships among those important structures; 3. what is the differences between those controls for struct snd_kcontrol_new and struct snd_soc_dapm_widget; 4. Should those sinks, sources, paths be defined in the arrays of struct snd_dapm_widget; 5. How can I select those defined paths through ALSA user-mode library as I can't find any examples in those ALSA tutorials; Indeed, I'm right now trying to write a WM8985 driver for my s3c6400 based platform based on the kernel source v2.6.21 from Samsung. Thank you so much for your valuable advice and suggestions in advance. Best Regards, LEUNG, Chi Tat Senior Software Engineer CCT Tech Advanced Products Limited 18/F, CCT Telecom Building, 11 Wo Shing Street Fo Tan, Shatin, N.T., Hong Kong Tel: +852 26005276 Fax: +852 26948660 Email: ctleung(a)cct.com.hk Website: http://www.cct-tech.com.hk

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[alsa-devel] [PATCH 0/3 take2] ASoC: TWL4030: Handsfree pop removal and mute
by Peter Ujfalusi 25 May '09

25 May '09

Hello, Changes: REG_SW_SHADOW = 0x4A (instead of 0x50) The following series fixes the HandsfreeL/R pop removal sequence: The pop removal is moved from the MUX_E to PGA_E DAPM control and also the sequence is modified to match with the sequence described in the TRM. Note, that the power-down sequence is not described in the TRM, it has been implemented in a way that it should make sense. HandsetL/R mute: Since all bits are needed for the pop removal and they have to modified in a strict order/timing: I have added SW_SHADOW non HW register. This register at the moment only handles the HandsfreeL/R mute. --- Peter Ujfalusi (3): ASoC: TWL4030: Handsfree pop removal redesign ASoC: TWL4030: Add shadow register ASoC: TWL4030: HandsfreeL/R mute DAPM switch sound/soc/codecs/twl4030.c | 99 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/twl4030.h | 7 +++- 2 files changed, 85 insertions(+), 21 deletions(-)

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[alsa-devel] [PATCH V2 0/9] mpc5200 audio rework for AC97
by Jon Smirl 25 May '09

25 May '09

The following series implements audio support for the mpc5200. It adds an AC97 driver and STAC9766 codec driver. Board support for the Efika and Phytec pcm030 are also included. Mark is not enthused about soc-of-simple.c so rather than extend it for AC97 I altered the drivers to not use it. Instead they use the old way of manually binding everything. Mark would like to see OF binding more closely integrated to the core. Once a proper solution for OF binding is agreed upon it is easy to convert the existing drivers. A first step would be converting the existing codec drivers so that they can be dynamically loaded. Grant, please check over the spin locking on register access. I'm not clear on why and when the registers have to be protected. Once these basic drivers are in-kernel and more people are testing them, I can add more features like pause/resume, power management, etc based on feedback. I2S will get examined in more detail for the next kernel cycle. Our multi-channel prototype I2S hardware is just about working. Once I get the multi-channel hardware I will implement and heavily test a lot more I2S capability. --- Jon Smirl (9): Support for AC97 on Phytec pmc030 base board. Fabric bindings for STAC9766 on the Efika AC97 driver for mpc5200 Codec for STAC9766 used on the Efika Main rewite of the mpc5200 audio DMA code Add a few more mpc5200 PSC defines Rename the PSC functions to DMA Basic split of mpc5200 DMA code out from mpc5200_psc_i2s Register the wm9712 DAIs on module load sound/soc/fsl/Kconfig | 31 + sound/soc/fsl/Makefile | 7 sound/soc/fsl/efika-audio-fabric.c | 94 ++++ sound/soc/fsl/mpc5200_dma.c | 635 ++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_dma.h | 80 ++++ sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 750 ++--------------------------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 94 ++++ 10 files changed, 1414 insertions(+), 698 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_dma.c create mode 100644 sound/soc/fsl/mpc5200_dma.h create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH 1/2] pxa2xx-ac97-lib: move set_resetgpio to pxa27x.c
by Dmitry Eremin-Solenikov 25 May '09

25 May '09

The code related to set_resetgpio is pxa27x specific and should not be present in sound driver lib. Move it to appropriate place. The code is moved to arch/arm as it will be converted from (exported) pxa_gpio_mode() calls to use MFP which are unexported to modules. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/mach-pxa/include/mach/audio.h | 24 ++++++++++++++++ arch/arm/mach-pxa/pxa27x.c | 37 +++++++++++++++++++++++++ sound/arm/pxa2xx-ac97-lib.c | 47 +++----------------------------- 3 files changed, 65 insertions(+), 43 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/audio.h b/arch/arm/mach-pxa/include/mach/audio.h index 16eb025..517c868 100644 --- a/arch/arm/mach-pxa/include/mach/audio.h +++ b/arch/arm/mach-pxa/include/mach/audio.h @@ -24,4 +24,28 @@ typedef struct { extern void pxa_set_ac97_info(pxa2xx_audio_ops_t *ops); +/* + * Beware PXA27x bugs: + * + * o Slot 12 read from modem space will hang controller. + * o CDONE, SDONE interrupt fails after any slot 12 IO. + * + * We therefore have an hybrid approach for waiting on SDONE (interrupt or + * 1 jiffy timeout if interrupt never comes). + */ + +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +#if defined(CONFIG_PXA27x) +extern void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action); +#else +static inline void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action) +{ +} +#endif + #endif diff --git a/arch/arm/mach-pxa/pxa27x.c b/arch/arm/mach-pxa/pxa27x.c index a425ec7..a0e787e 100644 --- a/arch/arm/mach-pxa/pxa27x.c +++ b/arch/arm/mach-pxa/pxa27x.c @@ -28,6 +28,7 @@ #include <mach/pm.h> #include <mach/dma.h> #include <mach/i2c.h> +#include <mach/audio.h> #include "generic.h" #include "devices.h" @@ -345,6 +346,42 @@ void __init pxa27x_set_i2c_power_info(struct i2c_pxa_platform_data *info) pxa_register_device(&pxa27x_device_i2c_power, info); } +/** + * pxa27x_ac97_reset - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +/* temporary include */ +#include <mach/pxa2xx-gpio.h> +void pxa27x_ac97_reset(int reset_gpio, int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_ALT_FN_2_OUT; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} +EXPORT_SYMBOL(pxa27x_ac97_reset); + static struct platform_device *devices[] __initdata = { &pxa27x_device_udc, &pxa_device_ffuart, diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fdca97..2227314 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -42,45 +42,6 @@ static int reset_gpio; * 1 jiffy timeout if interrupt never comes). */ -enum { - RESETGPIO_FORCE_HIGH, - RESETGPIO_FORCE_LOW, - RESETGPIO_NORMAL_ALTFUNC -}; - -/** - * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA - * @mode: chosen action - * - * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line - * must be done to insure proper work of AC97 reset line. This function - * computes the correct gpio_mode for further use by reset functions, and - * applied the change through pxa_gpio_mode. - */ -static void set_resetgpio_mode(int resetgpio_action) -{ - int mode = 0; - - if (reset_gpio) - switch (resetgpio_action) { - case RESETGPIO_NORMAL_ALTFUNC: - if (reset_gpio == 113) - mode = 113 | GPIO_ALT_FN_2_OUT; - if (reset_gpio == 95) - mode = 95 | GPIO_ALT_FN_1_OUT; - break; - case RESETGPIO_FORCE_LOW: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; - break; - case RESETGPIO_FORCE_HIGH: - mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; - break; - }; - - if (mode) - pxa_gpio_mode(mode); -} - unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -176,10 +137,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - set_resetgpio_mode(RESETGPIO_FORCE_HIGH); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -353,7 +314,7 @@ int pxa2xx_ac97_hw_resume(void) } if (cpu_is_pxa27x()) { /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -395,7 +356,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); + pxa27x_ac97_reset(reset_gpio, RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- 1.6.2.4

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Mario Contreras 25 May '09

25 May '09

Hello, I have this card. Will the unstable version work with my card? I have read it has a different chipset. I installed it a few days ago, but there wasn't any sound. I have Ubuntu 9.04. When I test it in sound preferences it freezes. I would like to test it. If I can be helpful, I'd appreciate you help me. Than you.

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[alsa-devel] [PATCH 0/3] ASoC: TWL4030: Handsfree pop removal and mute
by Peter Ujfalusi 25 May '09

25 May '09

Hello, The following series fixes the HandsfreeL/R pop removal sequence: The pop removal is moved from the MUX_E to PGA_E DAPM control and also the sequence is modified to match with the sequence described in the TRM. Note, that the power-down sequence is not described in the TRM, it has been implemented in a way that it should make sense. HandsetL/R mute: Since all bits are needed for the pop removal and they have to modified in a strict order/timing: I have added SW_SHADOW non HW register. This register at the moment only handles the HandsfreeL/R mute. --- Peter Ujfalusi (3): ASoC: TWL4030: Handsfree pop removal redesign ASoC: TWL4030: Add shadow register ASoC: TWL4030: HandsfreeL/R mute DAPM switch sound/soc/codecs/twl4030.c | 99 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/twl4030.h | 7 +++- 2 files changed, 85 insertions(+), 21 deletions(-)

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[alsa-devel] snd_pcm_writei() return value
by Cristian Morales Vega 25 May '09

25 May '09

Hi, I have an stupid doubt... The snd_pcm_writei() documentation says: "If the blocking behaviour is selected, then routine waits until all requested bytes are played or put to the playback ring buffer. The count of bytes can be less only if a signal or underrun occurred." and at the same time it says "-EPIPE an underrun occurred" is a possible return value. Then snd_pcm_recover() documentation says: "This functions handles -EINTR (interrupted system call), -EPIPE (overrun or underrun) and -ESTRPIPE (stream is suspended) error codes trying to prepare given stream for next I/O." So if I want to write the full content of a buffer, would this be valid?: while(buffer.length > 0) { snd_pcm_sframes_t written = snd_pcm_writei(device.handle, buffer.data, buffer.length); if(written < 0) snd_pcm_recover(device.handle, written, 1); else buffer.length = 0; } "The count of bytes can be less only if a signal ... occurred." makes me think that if a signal interrups snd_pcm_writei() it will return a value between 0 and buffer.length, but "This functions handles -EINTR (interrupted system call)" makes me think that it will always return -EINTR. "The count of bytes can be less only if a ... underrun occurred.", even if technically correct, also seems to be against the "-EPIPE an underrun occurred" return value. So, in the previous code, "if(written < 0)" should be changed by "if(written < buffer.length)"? Anyway, how snd_pcm_recover() handles an -EINTR? If only half the data has been written it can't modify buffer.data to point to the last sample written+1. Perhaps I need a full: uint32_t *buffer_ptr = buffer.data; while(buffer.length > 0) { snd_pcm_sframes_t written = snd_pcm_writei(device.handle, buffer_ptr, buffer.length); if(written < 0) { snd_pcm_recover(device.handle, written, 1); } else if(written <= buffer.length) { buffer.length -= written; buffer_ptr += written; } } ? And still, in this case an interrupt would be handled by snd_pcm_recover() (written < 0) or by the "else" code? Thanks.

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[alsa-devel] [PATCH] ALSA: hda - Add forced codec-slots for ASUS W5Fm
by Ozan Çağlayan 25 May '09

25 May '09

ASUS W5Fm needs the fixed codec-slots to probe to override the BIOS problem like W5F. Tested-by: Alp Kılıç <kilic.alp(a)gmail.com> Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 21e99cf..4419e11 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2142,6 +2142,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), /* forced codec slots */ SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), {} }; -- 1.6.2

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Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985
by Leung Chi Tat 25 May '09

25 May '09

Thank you so much for your suggestion. I've already copied a codec source for wm8980 from linux-2.6-ASoC main trunk. I have already to modify it back to the old api's and old structures for kernel v2.6.21. What remains I can't get through are those points I listed in my previous email. Could you give me some hints? Best Regards, LEUNG, Chi Tat Senior Software Engineer CCT Tech Advanced Products Limited 18/F, CCT Telecom Building, 11 Wo Shing Street Fo Tan, Shatin, N.T., Hong Kong Tel: +852 26005276 Fax: +852 26948660 Email: ctleung(a)cct.com.hk Website: http://www.cct-tech.com.hk -----Original Message----- From: Jon Smirl [mailto:jonsmirl@gmail.com] Sent: Monday, May 25, 2009 11:45 AM To: Leung Chi Tat Cc: alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] Help on writing an ALSA ASoC Driver for WM8985 On Sun, May 24, 2009 at 11:31 PM, Leung Chi Tat <ctleung(a)cct.com.hk> wrote: > Hi all, > > I'm new to ALSA ASoC and ALSA. I'd like get some advice where I can get more information about writing an ASoC driver. I've browsed the internet couples of days and I can only find links on ALSA driver api's and writing an ALSA driver for PCI devices. > > It seems there is not much information on how to writing an ALSA ASoC driver. I'm stuck on the following questions: > 1. describing those important structures, e.g. struct soc_enum, struct snd_kcontrol_new, struct snd_soc_dapm_widget; > 2. what is/are the relationships among those important structures; > 3. what is the differences between those controls for struct snd_kcontrol_new and struct snd_soc_dapm_widget; > 4. Should those sinks, sources, paths be defined in the arrays of struct snd_dapm_widget; > 5. How can I select those defined paths through ALSA user-mode library as I can't find any examples in those ALSA tutorials; > > Indeed, I'm right now trying to write a WM8985 driver for my s3c6400 based platform based on the kernel source v2.6.21 from Samsung. Look in sound/soc/codecs. There are implementations for many similar Wolfson chips. Just cut and paste them together to make the wm8985. > > Thank you so much for your valuable advice and suggestions in advance. > > Best Regards, > LEUNG, Chi Tat > Senior Software Engineer > CCT Tech Advanced Products Limited > 18/F, CCT Telecom Building, 11 Wo Shing Street > Fo Tan, Shatin, N.T., Hong Kong > Tel: +852 26005276 Fax: +852 26948660 > Email: ctleung(a)cct.com.hk > Website: http://www.cct-tech.com.hk > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by linux user 24 May '09

24 May '09

Hi I am running 2.6.28-11-generic (Kubunti install) with M4A78T-E Motherboard and also have no digital spdif output. aplay -L default:CARD=SB HDA ATI SB, VT1708S Analog Default Audio Device front:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog Front speakers surround40:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=SB,DEV=0 HDA ATI SB, VT1708S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) hdmi:CARD=HDMI HDA ATI HDMI, ATI HDMI HDMI Audio Output I have read this thread ( http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/016958.html ) and gather that it is a problem with the BIOS and that it was fixable by patching alsa source. As a newbie and not wanting to destroy my alsa source files can you explain or refer me to a good howto of the steps needed? Thanks http://mailman.alsa-project.org/pipermail/alsa-devel/2009-May/016958.html

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Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.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. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * 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/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.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. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * 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/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.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. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * 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/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
by Demian Martin 24 May '09

24 May '09

Demian Martin PDS 209 613 6990 -----Original Message----- From: alsa-devel-request(a)alsa-project.org Date: Sun, 24 May 2009 01:13:19 To: <alsa-devel(a)alsa-project.org> Subject: Alsa-devel Digest, Vol 27, Issue 124 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl) 2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl) 3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl) ---------------------------------------------------------------------- Message: 1 Date: Sat, 23 May 2009 19:13:03 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC defines To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231303.17919.35877.stgit@terra> Content-Type: text/plain; charset="utf-8" Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree. Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h index a218da6..fb84120 100644 --- a/arch/powerpc/include/asm/mpc52xx_psc.h +++ b/arch/powerpc/include/asm/mpc52xx_psc.h @@ -28,6 +28,10 @@ #define MPC52xx_PSC_MAXNUM 6 /* Programmable Serial Controller (PSC) status register bits */ +#define MPC52xx_PSC_SR_UNEX_RX 0x0001 +#define MPC52xx_PSC_SR_DATA_VAL 0x0002 +#define MPC52xx_PSC_SR_DATA_OVR 0x0004 +#define MPC52xx_PSC_SR_CMDSEND 0x0008 #define MPC52xx_PSC_SR_CDE 0x0080 #define MPC52xx_PSC_SR_RXRDY 0x0100 #define MPC52xx_PSC_SR_RXFULL 0x0200 @@ -61,6 +65,12 @@ #define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001 /* PSC interrupt status/mask bits */ +#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001 +#define MPC52xx_PSC_IMR_DATA_VALID 0x0002 +#define MPC52xx_PSC_IMR_DATA_OVR 0x0004 +#define MPC52xx_PSC_IMR_CMD_SEND 0x0008 +#define MPC52xx_PSC_IMR_ERROR 0x0040 +#define MPC52xx_PSC_IMR_DEOF 0x0080 #define MPC52xx_PSC_IMR_TXRDY 0x0100 #define MPC52xx_PSC_IMR_RXRDY 0x0200 #define MPC52xx_PSC_IMR_DB 0x0400 @@ -117,6 +127,7 @@ #define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24) #define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24) +#define MPC52xx_PSC_SICR_AWR (1 << 30) #define MPC52xx_PSC_SICR_GENCLK (1 << 23) #define MPC52xx_PSC_SICR_I2S (1 << 22) #define MPC52xx_PSC_SICR_CLKPOL (1 << 21) ------------------------------ Message: 2 Date: Sat, 23 May 2009 19:13:07 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the Efika To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231307.17919.5277.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 codec for STAC9766 used on the Efika. Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- 0 files changed, 0 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7f78b65..cb07d9b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C + select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C @@ -93,6 +94,9 @@ config SND_SOC_PCM3008 config SND_SOC_SSM2602 tristate +config SND_SOC_STAC9766 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c55fa..46c007c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c new file mode 100644 index 0000000..7740cd5 --- /dev/null +++ b/sound/soc/codecs/stac9766.c @@ -0,0 +1,470 @@ +/* + * stac9766.c -- ALSA SoC STAC9766 codec support + * + * Copyright 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.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. + * + * Features:- + * + * o Support for AC97 Codec, S/PDIF + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-of-simple.h> + +#include "stac9766.h" + +#define STAC9766_VERSION "0.10" + +/* + * STAC9766 register cache + */ +static const u16 stac9766_reg[] = { + 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */ + 0x0000, 0x0000, 0x8008, 0x8008, /* e */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ + 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ + 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */ + 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */ + 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */ + 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ + 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */ +}; + +static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; +static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; +static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; +static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; +static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; +static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; +static const char *stac9766_boost1[] = {"0dB", "10dB"}; +static const char *stac9766_boost2[] = {"0dB", "20dB"}; +static const char *stac9766_stereo_mic[] = {"Off", "On"}; + +static const struct soc_enum stac9766_record_enum = + SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); +static const struct soc_enum stac9766_mono_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); +static const struct soc_enum stac9766_mic_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); +static const struct soc_enum stac9766_SPDIF_enum = + SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); +static const struct soc_enum stac9766_popbypass_enum = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); +static const struct soc_enum stac9766_record_all_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux); +static const struct soc_enum stac9766_boost1_enum = + SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ +static const struct soc_enum stac9766_boost2_enum = + SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ +static const struct soc_enum stac9766_stereo_mic_enum = + SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); + +static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); +static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); +static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0); +static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200); + +static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { + SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv), + SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv), + SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1), + + SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv), + SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1), + + + SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv), + SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1), + SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv), + SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), + + SOC_ENUM("Mic Boost1", stac9766_boost1_enum), + SOC_ENUM("Mic Boost2", stac9766_boost2_enum), + SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv), + SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), + SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum), + + SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), + SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1), + SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), + SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), + + SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv), + SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), + SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1), + SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), + + SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum), + SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum), + SOC_ENUM("Record All Mux", stac9766_record_all_enum), + SOC_ENUM("Record Mux", stac9766_record_enum), + SOC_ENUM("Mono Mux", stac9766_mono_enum), + SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum), +}; + +int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + soc_ac97_ops.write(codec->ac97, reg, val); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return 0; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + soc_ac97_ops.write(codec->ac97, reg, val); + cache[reg / 2] = val; + return 0; +} + +unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 val = 0, *cache = codec->reg_cache; + + if (reg > AC97_STAC_PAGE0) { + stac9766_ac97_write(codec, AC97_INT_PAGING, 0); + val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0); + stac9766_ac97_write(codec, AC97_INT_PAGING, 1); + return val; + } + if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + return -EIO; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || + reg == AC97_VENDOR_ID2) { + + val = soc_ac97_ops.read(codec->ac97, reg); + return val; + } + return cache[reg / 2]; +} + +static int ac97_analog_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + + vra |= 0x1; /* enable variable rate audio */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned short reg, vra; + + stac9766_ac97_write(codec, AC97_SPDIF, 0x2002); + + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra |= 0x5; /* Enable VRA and SPDIF out */ + + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + + reg = AC97_PCM_FRONT_DAC_RATE; + + return stac9766_ac97_write(codec, reg, runtime->rate); +} + +static int ac97_digital_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned short vra; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); + vra &= !0x04; + stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); + break; + } + return 0; +} + +static int stac9766_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: /* full On */ + case SND_SOC_BIAS_PREPARE: /* partial On */ + case SND_SOC_BIAS_STANDBY: /* Off, with power */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SND_SOC_BIAS_OFF: /* Off, without power */ + /* disable everything including AC link */ + stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +int stac9766_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); + if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) + return -EIO; + return 0; +} + +static int stac9766_codec_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int stac9766_codec_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 id, reset; + + reset = 0; + /* give the codec an AC97 warm reset to start the link */ +reset: + if (reset > 5) { + printk(KERN_ERR "stac9766 failed to resume"); + return -EIO; + } + codec->ac97->bus->ops->warm_reset(codec->ac97); + id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); + if (id != 0x4c13) { + stac9766_reset(codec, 0); + reset++; + goto reset; + } + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_dai_ops stac9766_dai_ops_analog = +{ + .prepare = ac97_analog_prepare, +}; + +static struct snd_soc_dai_ops stac9766_dai_ops_digital = +{ + .prepare = ac97_digital_prepare, + .trigger = ac97_digital_trigger, +}; + +struct snd_soc_dai stac9766_dai[] = { +{ + .name = "stac9766 analog", + .id = 0, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + .capture = { + .stream_name = "stac9766 analog", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SND_SOC_STD_AC97_FMTS, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_analog, +}, +{ + .name = "stac9766 IEC958", + .id = 1, + .ac97_control = 1, + + /* stream cababilities */ + .playback = { + .stream_name = "stac9766 IEC958", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE, + }, + /* alsa ops */ + .ops = &stac9766_dai_ops_digital, +}}; +EXPORT_SYMBOL_GPL(stac9766_dai); + +static int stac9766_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(stac9766_reg); + codec->reg_cache_step = 2; + + codec->name = "STAC9766"; + codec->owner = THIS_MODULE; + codec->dai = stac9766_dai; + codec->num_dai = ARRAY_SIZE(stac9766_dai); + codec->write = stac9766_ac97_write; + codec->read = stac9766_ac97_read; + codec->set_bias_level = stac9766_set_bias_level; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + stac9766_reset(codec, 0); + ret = stac9766_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); + goto reset_err; + } + + stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE( + stac9766_snd_ac97_controls)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->private_data); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int stac9766_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_stac9766 = +{ + .probe = stac9766_codec_probe, + .remove = stac9766_codec_remove, + .suspend = stac9766_codec_suspend, + .resume = stac9766_codec_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); + +static int __init stac9766_modinit(void) +{ + return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_init(stac9766_modinit); + +static void __exit stac9766_exit(void) +{ + snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); +} +module_exit(stac9766_exit); + +MODULE_DESCRIPTION("ASoC stac9766 driver"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h new file mode 100644 index 0000000..65642eb --- /dev/null +++ b/sound/soc/codecs/stac9766.h @@ -0,0 +1,21 @@ +/* + * stac9766.h -- STAC9766 Soc Audio driver + */ + +#ifndef _STAC9766_H +#define _STAC9766_H + +#define AC97_STAC_PAGE0 0x1000 +#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A) +#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E) +#define AC97_STAC_STEREO_MIC 0x78 + +/* STAC9766 DAI ID's */ +#define STAC9766_DAI_AC97_ANALOG 0 +#define STAC9766_DAI_AC97_DIGITAL 1 + +extern struct snd_soc_dai stac9766_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_stac9766; + + +#endif ------------------------------ Message: 3 Date: Sat, 23 May 2009 19:13:09 -0400 From: Jon Smirl <jonsmirl(a)gmail.com> Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200 To: grant.likely(a)secretlab.ca, linuxppc-dev(a)ozlabs.org, alsa-devel(a)alsa-project.org, broonie(a)sirena.org.uk Message-ID: <20090523231309.17919.83073.stgit@terra> Content-Type: text/plain; charset="utf-8" AC97 driver for mpc5200 Signed-off-by: Jon Smirl <jonsmirl(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 1 sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + 4 files changed, 421 insertions(+), 0 deletions(-) create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 1918c78..3bce952 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S select PPC_BESTCOMM_GEN_BD help Say Y here to support the MPC5200 PSCs in I2S mode. + +config SND_SOC_MPC5200_AC97 + tristate "Freescale MPC5200 PSC in AC97 mode driver" + depends on PPC_MPC52xx && PPC_BESTCOMM + select AC97_BUS + select SND_MPC52xx_DMA + select PPC_BESTCOMM_GEN_BD + help + Say Y here to support the MPC5200 PSCs in AC97 mode. + + diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7731ef2..14631a1 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o +obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c new file mode 100644 index 0000000..fa1bb9a --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -0,0 +1,394 @@ +/* + * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip. + * + * Copyright (C) 2009 Jon Smirl, Digispeaker + * Author: Jon Smirl <jonsmirl(a)gmail.com> + * + * 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/module.h> +#include <linux/of_device.h> +#include <linux/of_platform.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mpc52xx_psc.h> + +#include "mpc5200_dma.h" +#include "mpc5200_psc_ac97.h" + +#define DRV_NAME "mpc5200-psc-ac97" + +/* ALSA only supports a single AC97 device so static is recommend here */ +static struct psc_dma *psc_dma; + +static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + int timeout; + unsigned int val; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 bus (rdy)\n"); + return 0xffff; + } + + /* Do the read */ + out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); + + /* Wait for the answer */ + timeout = 1000; + while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_DATA_VAL) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + return 0xffff; + } + + /* Get the data */ + val = in_be32(&psc_dma->psc_regs->ac97_data); + if ( ((val>>24) & 0x7f) != reg ) { + pr_err("reg echo error on ac97 read\n"); + return 0xffff; + } + val = (val >> 8) & 0xffff; + + spin_unlock(&psc_dma->lock); + return (unsigned short) val; +} + +static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) +{ + int timeout; + + spin_lock(&psc_dma->lock); + + /* Wait for it to be ready */ + timeout = 1000; + while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) & + MPC52xx_PSC_SR_CMDSEND) ) + udelay(10); + + if (!timeout) { + pr_err("timeout on ac97 write\n"); + return; + } + + /* Write data */ + out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + spin_unlock(&psc_dma->lock); +} + +static void psc_ac97_cold_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + /* Do a cold reset */ + out_8(&regs->op1, MPC52xx_PSC_OP_RES); + udelay(10); + out_8(&regs->op0, MPC52xx_PSC_OP_RES); + udelay(50); + + /* PSC recover from cold reset (cfr user manual, not sure if useful) */ + out_be32(&regs->sicr, in_be32(&regs->sicr)); +} + +static void psc_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; + + out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR); + udelay(3); + out_be32(&regs->sicr, psc_dma->sicr); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = psc_ac97_read, + .write = psc_ac97_write, + .reset = psc_ac97_cold_reset, + .warm_reset = psc_ac97_warm_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +#ifdef CONFIG_PM +static int psc_ac97_suspend(struct snd_soc_dai *dai) +{ + return 0; +} + +static int psc_ac97_resume(struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define psc_ac97_suspend NULL +#define psc_ac97_resume NULL +#endif + +static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" + " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" + " rate=%i format=%i\n", + __func__, substream, params_period_size(params), + params_period_bytes(params), params_periods(params), + params_buffer_size(params), params_buffer_bytes(params), + params_channels(params), params_rate(params), params_format(params)); + + + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (params_channels(params) == 1) + psc_dma->slots |= 0x00000100; + else + psc_dma->slots |= 0x00000300; + } else { + if (params_channels(params) == 1) + psc_dma->slots |= 0x01000000; + else + psc_dma->slots |= 0x03000000; + } + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + spin_lock(&psc_dma->lock); + if (params_channels(params) == 1) + out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); + else + out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000); + spin_unlock(&psc_dma->lock); + + return 0; +} + +static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + psc_dma->slots &= 0xFFFF0000; + else + psc_dma->slots &= 0x0000FFFF; + + spin_lock(&psc_dma->lock); + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + spin_unlock(&psc_dma->lock); + break; + } + return 0; +} + +/* --------------------------------------------------------------------- + * ALSA SoC Bindings + * + * - Digital Audio Interface (DAI) template + * - create/destroy dai hooks + */ + +/** + * psc_ac97_dai_template: template CPU Digital Audio Interface + */ +static struct snd_soc_dai_ops psc_ac97_analog_ops = { + .hw_params = psc_ac97_hw_analog_params, + .trigger = psc_ac97_trigger, +}; + +static struct snd_soc_dai_ops psc_ac97_digital_ops = { + .hw_params = psc_ac97_hw_digital_params, +}; + +struct snd_soc_dai psc_ac97_dai[] = { +{ + .name = "AC97", + .suspend = psc_ac97_suspend, + .resume = psc_ac97_resume, + .playback = { + .channels_min = 1, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_BE, + }, + .ops = &psc_ac97_analog_ops, +}, +{ + .name = "SPDIF", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, + }, + .ops = &psc_ac97_digital_ops, +}}; +EXPORT_SYMBOL_GPL(psc_ac97_dai); + + + +/* --------------------------------------------------------------------- + * OF platform bus binding code: + * - Probe/remove operations + * - OF device match table + */ +static int __devinit psc_ac97_of_probe(struct of_device *op, + const struct of_device_id *match) +{ + int rc, i, id1, id2, timeout, max_reset; + struct snd_ac97 ac97; + struct mpc52xx_psc __iomem *regs; + + rc = mpc5200_audio_dma_create(op); + if (rc != 0) + return rc; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].dev = &op->dev; + + rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai)); + if (rc != 0) { + pr_err("Failed to register DAI\n"); + return 0; + } + + psc_dma = dev_get_drvdata(&op->dev); + regs = psc_dma->psc_regs; + ac97.private_data = psc_dma; + + for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++) + psc_ac97_dai[i].private_data = psc_dma; + + psc_dma->imr = 0; + out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr); + + /* Configure the serial interface mode to AC97 */ + psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97; + out_be32(&regs->sicr, psc_dma->sicr); + + /* No slots active */ + out_be32(&regs->ac97_slots, 0x00000000); + + /* AC97 clock is generated by the codec. + * Ensure that it starts ticking after codec reset. + */ + max_reset = 0; +reset: + if (max_reset++ > 5) { + dev_err(&op->dev, "AC97 codec failed to reset\n"); + mpc5200_audio_dma_destroy(op); + return -ENODEV; + } + + psc_ac97_cold_reset(&ac97); + psc_ac97_warm_reset(&ac97); + + /* first make sure it is low */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0) { + udelay(1); + if (timeout++ > 1000) + goto reset; + } + /* then wait for the transition to high */ + timeout = 0; + while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0) { + udelay(1); + if (timeout++ > 1000) + psc_ac97_warm_reset(&ac97); + } + + /* Go */ + out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE); + + id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1); + id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2); + + dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2); + + return 0; +} + +static int __devexit psc_ac97_of_remove(struct of_device *op) +{ + return mpc5200_audio_dma_destroy(op); +} + +/* Match table for of_platform binding */ +static struct of_device_id psc_ac97_match[] __devinitdata = { + { .compatible = "fsl,mpc5200-psc-ac97", }, + { .compatible = "fsl,mpc5200b-psc-ac97", }, + {} +}; +MODULE_DEVICE_TABLE(of, psc_ac97_match); + +static struct of_platform_driver psc_ac97_driver = { + .match_table = psc_ac97_match, + .probe = psc_ac97_of_probe, + .remove = __devexit_p(psc_ac97_of_remove), + .driver = { + .name = "mpc5200-psc-ac97", + .owner = THIS_MODULE, + }, +}; + +/* --------------------------------------------------------------------- + * Module setup and teardown; simply register the of_platform driver + * for the PSC in AC97 mode. + */ +static int __init psc_ac97_init(void) +{ + return of_register_platform_driver(&psc_ac97_driver); +} +module_init(psc_ac97_init); + +static void __exit psc_ac97_exit(void) +{ + of_unregister_platform_driver(&psc_ac97_driver); +} +module_exit(psc_ac97_exit); + +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_DESCRIPTION("mpc5200 AC97 module"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h new file mode 100644 index 0000000..4bc18c3 --- /dev/null +++ b/sound/soc/fsl/mpc5200_psc_ac97.h @@ -0,0 +1,15 @@ +/* + * Freescale MPC5200 PSC in AC97 mode + * ALSA SoC Digital Audio Interface (DAI) driver + * + */ + +#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ +#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ + +extern struct snd_soc_dai psc_ac97_dai[]; + +#define MPC5200_AC97_NORMAL 0 +#define MPC5200_AC97_SPDIF 1 + +#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */ ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 27, Issue 124 *******************************************

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Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G
by Karthik Ramgopal 23 May '09

23 May '09

Hi, Even after enabling DEBUG using --with-debug option with configure when compiling the alsa-driver snapshot leaves no more error messages in the dmesg logs. The only message relating to HDA is: [ 41.104817] hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. I just got the line-out output to work by toggling the surround volume. I still think it's a mapping problem. The headphone switch is turning the subwoofer on and off, and the surround volume is controlling the line-out volume. None of the line-in devices function and all of the amp-ins 0x0b to 0x0f are muted. Cannot find any verb to unmute them. hda-verb with SET_EAPD_BTLENABLE 2 at 0x15 is still needed at startup to fire up the amps. Please help. Regards, Karthik

3 21

[alsa-devel] Testers wanted: New SB X-Fi driver
by Ge-org Brohammer 23 May '09

23 May '09

Good day I compiled and installed alsa-lib-1.0.20 and alsa-utils-1.0.20 and then your driver on ubuntu Jaunty 64 bit. I have the X-FI Music PCI card and it seems to split the outputs into separate cards? This is the output from aplay -l **** List of PLAYBACK Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: Front/WaveIn [Front/WaveIn] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 1: Rear [Rear] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 2: Center/LFE [Center/LFE] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 3: Surround [Surround] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: XFi [Creative X-Fi], device 4: IEC958 Non-audio [IEC958 Non-audio] Subdevices: 1/1 Subdevice #0: subdevice #0 This a problem for me because I cannot play all the channels at once either front side rear, etc... Let me know if any extra info is required. Regards

1 0

[alsa-devel] force byte-swap at soc level before codec?
by Brad Parker 23 May '09

23 May '09

Hi I'm an ALSA SOC newbie. I created a new SOC driver for a new ARM chip but I'm having endian issues at the codec. I started with the pxa soc code since I am failure with the pxa* i2s hardware. The new code is working and data flows fine but due to hardware constraints the bytes arrive at the codec swapped. The codec claims S16_LE. The i2s layer claims S16_LE as does the pcm layer inside soc. The source data in user land is S16_LE. But when the i2s hardware (which I can't change) does dma the bytes go to the codec in reverse order. So, I need a byte swap somewhere. I verified this by sending constant data from user land via the oss compat layer (i.e. /dev/dsp). If I swap them in userland they come out on the scope at the codec in the correct order. Any know the simplest way to do this? I've tried a bunch of ways but nothing seems to work correctly. It seems like ALSA already handles this as long as I tell the right thing from below. -brad

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[alsa-devel] [PATCH 1/1] hdsp: right firmware path
by Raphaël Doursenaud 23 May '09

23 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 From: Raphaël Doursenaud <rdoursenaud(a)free.fr> Correct firmware path to reflect actual location. This was causing hotplug to fail. Inspired by the emu10k1 module. Signed-off-by: Raphael Doursenaud <rdoursenaud(a)free.fr> - --- sound/pci/rme9652/hdsp.c | 22 ++++++++++++++-------- 1 files changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 314e735..48f5434 100644 - --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -59,11 +59,17 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP}," "{RME HDSP-9652}," "{RME HDSP-9632}}"); + +#define MULTIFACE_FILENAME "hdsploader/multiface_firmware.bin" +#define MULTIFACE_REV11_FILENAME "hdsploader/multiface_firmware_rev11.bin" +#define DIGIFACE_FILENAME "hdsploader/digiface_firmware.bin" +#define DIGIFACE_REV11_FILENAME "hdsploader/digiface_firmware_rev11.bin" + #ifdef HDSP_FW_LOADER - -MODULE_FIRMWARE("multiface_firmware.bin"); - -MODULE_FIRMWARE("multiface_firmware_rev11.bin"); - -MODULE_FIRMWARE("digiface_firmware.bin"); - -MODULE_FIRMWARE("digiface_firmware_rev11.bin"); +MODULE_FIRMWARE(MULTIFACE_FILENAME); +MODULE_FIRMWARE(MULTIFACE_REV11_FILENAME); +MODULE_FIRMWARE(DIGIFACE_FILENAME); +MODULE_FIRMWARE(DIGIFACE_REV11_FILENAME); #endif #define HDSP_MAX_CHANNELS 26 @@ -4928,15 +4934,15 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) switch (hdsp->io_type) { case Multiface: if (hdsp->firmware_rev == 0xa) - - fwfile = "multiface_firmware.bin"; + fwfile = MULTIFACE_FILENAME; else - - fwfile = "multiface_firmware_rev11.bin"; + fwfile = MULTIFACE_REV11_FILENAME; break; case Digiface: if (hdsp->firmware_rev == 0xa) - - fwfile = "digiface_firmware.bin"; + fwfile = DIGIFACE_FILENAME; else - - fwfile = "digiface_firmware_rev11.bin"; + fwfile = DIGIFACE_REV11_FILENAME; break; default: snd_printk(KERN_ERR "Hammerfall-DSP: invalid io_type %d\n", hdsp->io_type); - -- Raphaël Doursenaud -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoWurIACgkQaZKmNAdXaVVYYACdEyheX1+7sUbWURScDRFV66XV s4gAniZHnEfwRG012ZQmaDAhDhNXQf9Z =Ex8j -----END PGP SIGNATURE-----

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 22 May '09

22 May '09

The following changes since commit e24805dd85283ac0912b9c400768a4d171b400ff: Atsushi Nemoto (1): ASoC: Add TXx9 AC link controller driver (v3) 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 Mark Brown (2): ASoC: Make snd_soc_read() and snd_soc_write() functions ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier driver Peter Ujfalusi (5): ASoC: TWL4030: Change DAPM routings and controls for DACs and PGAs ASoC: TWL4030: Move the Headset pop-attenuation code to PGA event ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal ASoC: TWL4030: Differentiate the playback streams include/sound/soc.h | 17 +- include/sound/wm9081.h | 25 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 285 ++++++--- sound/soc/codecs/twl4030.h | 5 + sound/soc/codecs/wm9081.c | 1532 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9081.h | 787 +++++++++++++++++++++++ sound/soc/omap/sdp3430.c | 7 + 9 files changed, 2559 insertions(+), 105 deletions(-) create mode 100644 include/sound/wm9081.h create mode 100644 sound/soc/codecs/wm9081.c create mode 100644 sound/soc/codecs/wm9081.h

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[alsa-devel] bootup message "Control name ... truncated to ..." - what to fix?
by Andreas Mohr 22 May '09

22 May '09

Hi all, On my i8k, I'm getting this on bootup (2.6.30-rc5): ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phase Inversion Playback Switch' truncated to 'Sigmatel Surround Phase Inversion Playback ' I would quickly have submitted a patch for this (and similar items), problem is: - what to fix: control name or struct definition? - "Sigmatel Surround" might be a relatively important prefix, thus removing "Sigmatel " from relevant names possibly not a good idea andi@note:/usr/src/linux-2.6.30-rc5/sound$ find . -name "*.c" -o -name "*.h"|xargs grep "Sigmatel Surround" ./pci/ac97/ac97_patch.c:AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0); ./pci/ac97/ac97_patch.c: snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Sigmatel Surround Playback"); ./pci/ca0106/ca0106_mixer.c: "Sigmatel Surround Phase Inversion Playback ", ./pci/emu10k1/emu10k1x.c: * the PCM Playback Volume, Sigmatel Surround Playback Volume and sound/core/control.c/snd_ctl_new1(): if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); if (strcmp(ncontrol->name, kctl.id.name) != 0) snd_printk(KERN_WARNING "Control name '%s' truncated to '%s'\n", ncontrol->name, kctl.id.name); } include/sound/asound.h: struct snd_ctl_elem_id { unsigned int numid; /* numeric identifier, zero = invalid */ snd_ctl_elem_iface_t iface; /* interface identifier */ unsigned int device; /* device/client number */ unsigned int subdevice; /* subdevice (substream) number */ unsigned char name[44]; /* ASCII name of item */ unsigned int index; /* index of item */ }; I suppose (well, DAMN sure) that name is only 44 bytes due to exactly 64-byte cacheline-compatible struct alignment status. Note that status quo is pretty bad, since in another file already: ./pci/ca0106/ca0106_mixer.c: "Sigmatel Surround Phase Inversion Playback ", . IOW we already violate official mixer control name specs (by omitting "Switch" - see ControlNames.txt) simply in order to obey the 44-byte limit. Given that snd_ctl_elem_id 64 byte is a rather hard and arguably important limit, I'd suggest changing the names, but how? Unfortunately Sigmatel Surround PhaseInvert Playback Switch doesn't fit the bill either, since it's one char too long. Thus it will probably be Surround Phase Inversion Playback Switch (and thus changing "Sigmatel Surround" at all affected controls) Thanks, Andreas Mohr

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[alsa-devel] more on appl_ptr and DMA overrun at end of stream
by Jon Smirl 22 May '09

22 May '09

I've been reading Lennart's article on glitch free audio.... On my scatter/gather DMA hardware I generate an interrupt at the end of each period. On the last period this causes ALSA to issue a trigger(STOP). However, there is a latency from when that interrupt is generated and when ALSA does the trigger(STOP). During that latency my hardware starts playing the next buffer (which contains stale data). Playing stale data causes a noise burst. I fixed this by programming my DMA hardware to not overrun appl_ptr. There's another solution. ALSA already knows the stream ends mid-period and pads out the partial period with silence. If it simply padded out the next period with silence too the overrun problem would become non-audible and there would time to process trigger(STOP). If padding out two periods at end of stream is possible, it then becomes possible for me to turn off interrupts in my audio driver. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: change the streams names for playbacks
by Peter Ujfalusi 22 May '09

22 May '09

Hello, rename the HiFi and Voice playback streams to have unique names to aid DAPM to correctly pick the active playback streams on the twl4030 codec. --- Peter Ujfalusi (1): ASoC: TWL4030: Differentiate the playback streams sound/soc/codecs/twl4030.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-)

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[alsa-devel] [PATCH 0/2 take2] SoC: TWL4030 and SDP3430 HS ramp configuration
by Peter Ujfalusi 22 May '09

22 May '09

Hello, I have found few typos in the patches, resending... he following series adds support for platform dependent Headset pop removal configuration for the twl4030 codec. Platform drivers can fill the twl4030_setup_data structure and pass it with snd_soc_device->codec_data to the codec driver. As an example - based on the feedback for the ramp_delay value variations - the sdp3430.c board is updated to pass the correct ramp settings. If you know for the other platforms using the twl4030 codec the correct HS ramp configuration, please update those files... Mark: This series needs the DAPM restructuring and HS pop removal related patches for twl4030 codec. --- Peter Ujfalusi (2): ASoC: TWL4030: Add support for platform dependent configuration ASoC: SDP4030: Use the twl4030_setup_data for headset pop-removal sound/soc/codecs/twl4030.c | 22 +++++++++++++++++++--- sound/soc/codecs/twl4030.h | 5 +++++ sound/soc/omap/sdp3430.c | 7 +++++++ 3 files changed, 31 insertions(+), 3 deletions(-)

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[alsa-devel] Disabling HFL/R outputs in twl4030
by Lopez Cruz, Misael 22 May '09

22 May '09

It seems that there is no way to disable handsfree L/R outputs for twl4030 codec. That means that: - handsfree amplifiers are turned-on even if we are not interested in routing through handsfree - if the selected input of the handsfree mux is an active DAC (streaming) sound will be produced on hf outputs and the only way to silence them is routing from a different DAC, which doesn't seem like the right solution Currently, handsfree amplifiers are turned-on when the HandsfreeL/R muxes are powered by the framework. I was thinking in adding a dapm switch to select whether or not handsfree amplifiers should be powered, but the switch probably is not the correct approach to handle this situation. Maybe a regular kcontrol could be the answer. -Misa

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[alsa-devel] Releasing a handle without closing ?
by Werner Van Belle 22 May '09

22 May '09

Hello, I have an intruiging problem. I have two processes: a parent and a child. The parent forks off the child (as can be expected, the opposite would be somewhat unsettling), after it allocated and started to write to a pcm device. As can be expected, the child will also have a link to this dsp device. The problem now is that the child wants to release the pcm handle without closing it. Is there any particular function offering such possibility ? Wkr,- -- Dr. Werner Van Belle http://werner.yellowcouch.org/

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