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[alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Hello. I need to set proper period time for my application. What I need my code to do is to play sine wave with set frequency and for exact time of 1/1200 s (something about 834 us). When I try to set period time for 834 us I've got this error: Unable to set period time 834 for playback: Invalid argument . Honestly speaking I've got this error message no matter what value I try to set). The same situation take place when I try to set buffer time for something about 1000us (so one period could fit in my buffer). The purpose of this program is to generate sine wave for AFSK (audio frequency-shift keying). I want to play wave for period_time (834us) then if needed change frequency and play changed wave again for period_time with phase shifting. Wave generation with phase shifting for variable frequencies works, but not with period_time I need. Could anyone look at source and tell me why I can't set this values? I placed code with syntax highlighting at: http://pygments.org/demo/606/ . (Don't look at prepare_message function, it isn't used right now). Thanks from advice. Ignacy. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

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Re: [alsa-devel] installation steps for us-122l tascam driver
by belibem 18 Feb '08

18 Feb '08

Are there any news on the subject? What does it take to apply the patch? A packet for any major distribution would be greatly appreciated.

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[alsa-devel] Cirrus Logic CS42L52 Low power codec
by Brian Austin 18 Feb '08

18 Feb '08

Hello, I have a codec I would like to submit. The Cirrus Logic CS42L52 Low Power codec. I have the codec code as well as the machine code for the Cirrus Logic EP9407A, but cannot submit that yet. Is it OK to just submit the codec code for now? This does not use the new API, so a hint as to which tree to make the diff from would help. Thanks, Brian Austin

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[alsa-devel] Problem with DELL XPS M1530
by Travis Place 18 Feb '08

18 Feb '08

Hi all, Been helping a user on #alsa for a day or two now, with issues regarding recording from Mic on his Dell XPS M1530 laptop. He originally had alsa-driver 1.0.14, so i had him upgrade to 1.0.16. Now, his Mic works for recording, but he has lost playback on one of his HP jacks (the right one), and Internal laptop speakers. I had a brief look at the code in patch_sigmatel.c, to see what was going on.. But since im no code guru, i wasnt much help.. The output of alsa-info.sh is located at http://pastebin.ca/908579 I did notice some comments in the code about correcting HP and Mic jacks for certain models (atleast thats what I read it as). Any ideas would be appreciated. Travis (wishie). -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] unable to get alsa configured on Acer Aspire 1640Z
by Richard Lyons 18 Feb '08

18 Feb '08

Hi, I hope this is the right place to report this. Computer: laptop Acer Aspire 1640Z distro: Debian etch kernel: 2.6.17 lspci 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) lspci -nv 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 1025:009e Flags: bus master, fast devsel, latency 0, IRQ 177 Memory at d000c000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) I've tried model=acer and acer-aspire to no avail. I notice in dmesg after trying 'modprobe snd-hda-intel model=acer' this: hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... but alsamixer shows: Card: HDA Intel Chip: Realtek ALC883 at the top. There seems to be uncertainty about which model number, 882 or 883. I wonder if this is the source of the difficulty. HTH -- richard Lyons

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[alsa-devel] Intel8x0 8 channel sound
by Andrew Boettcher 18 Feb '08

18 Feb '08

For some intel 8x0 cards there is 8 channel sound (such as the ALC850 chip on NFORCE 4 boards), and this has not been patched yet though the patch has been on the bug tracker. Here it is again updated for 1.0.16. There is also a patch for the configuration in alsa-lib for cards/NFORCE.conf. I have been patching my sources every time I update my kernel, I figured it was time for someone to actually submit this somewhere else. The patch was written by Martin Ellis.

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[alsa-devel] alsa-info.sh script
by Travis Place 18 Feb '08

18 Feb '08

Is there any chance that i will be able to push updates for my script back to ALSA? Or is this to be a no-go ? Travis Place. (wishie) -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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[alsa-devel] [PATCH] Driver for Emagic Audiowerk 2
by Cédric Brégardis 18 Feb '08

18 Feb '08

Hi, We managed to write ALSA driver for Emagic Audiowerk 2 PCI sound card (based on Philips SAA7146 chip). You can find more information on https://gna.org/projects/aw2-alsa/. We think it's time to integrate the sources into the alsa/kernel tree. So, here is the patch for this driver. We hope this patch is usable. Regards, Jean-Christian Hassler and Cedric Bregardis Signed-off-by: Cedric Bregardis <cedric.bregardis(a)free.fr> Signed-off-by: Jean-Christian Hassler <jhassler(a)free.fr> --- sound/pci/Kconfig | 15 + sound/pci/Makefile | 1 + sound/pci/aw2/Makefile | 3 + sound/pci/aw2/SAA7146.h | 168 ++++++++++ sound/pci/aw2/aw2-alsa.c | 753 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.c | 613 +++++++++++++++++++++++++++++++++++ sound/pci/aw2/aw2-saa7146.h | 125 +++++++ sound/pci/aw2/aw2_tsl.h | 116 +++++++ 8 files changed, 1794 insertions(+), 0 deletions(-) create mode 100644 sound/pci/aw2/Makefile create mode 100644 sound/pci/aw2/SAA7146.h create mode 100644 sound/pci/aw2/aw2-alsa.c create mode 100644 sound/pci/aw2/aw2-saa7146.c create mode 100644 sound/pci/aw2/aw2-saa7146.h create mode 100644 sound/pci/aw2/aw2_tsl.h diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 812085d..194394b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -122,6 +122,21 @@ config SND_AU8830 To compile this driver as a module, choose M here: the module will be called snd-au8830. +config SND_AW2 + tristate "Emagic Audiowerk 2" + depends on SND + help + Say Y here to include support for Emagic Audiowerk 2 soundcards. + + Supported features: Analog and SPDIF output. Analog or SPDIF input. + Note: Switch between analog and digital input does not always work. + It can produce continuous noise. The workaround is to switch again + (and again) between digital and analog input until it works. + + To compile this driver as a module, choose M here: the module + will be called snd-aw2. + + config SND_AZT3328 tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" depends on SND && EXPERIMENTAL diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 2d42fd2..85ef14b 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -58,6 +58,7 @@ obj-$(CONFIG_SND) += \ ac97/ \ ali5451/ \ au88x0/ \ + aw2/ \ ca0106/ \ cs46xx/ \ cs5535audio/ \ diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile new file mode 100644 index 0000000..842335d --- /dev/null +++ b/sound/pci/aw2/Makefile @@ -0,0 +1,3 @@ +snd-aw2-objs := aw2-alsa.o aw2-saa7146.o + +obj-$(CONFIG_SND_AW2) += snd-aw2.o diff --git a/sound/pci/aw2/SAA7146.h b/sound/pci/aw2/SAA7146.h new file mode 100644 index 0000000..1fbc66e --- /dev/null +++ b/sound/pci/aw2/SAA7146.h @@ -0,0 +1,168 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +/* SAA7146 registers */ +#define PCI_BT_A 0x4C +#define IICTFR 0x8C +#define IICSTA 0x90 +#define BaseA1_in 0x94 +#define ProtA1_in 0x98 +#define PageA1_in 0x9C +#define BaseA1_out 0xA0 +#define ProtA1_out 0xA4 +#define PageA1_out 0xA8 +#define BaseA2_in 0xAC +#define ProtA2_in 0xB0 +#define PageA2_in 0xB4 +#define BaseA2_out 0xB8 +#define ProtA2_out 0xBC +#define PageA2_out 0xC0 +#define IER 0xDC +#define GPIO_CTRL 0xE0 +#define ACON1 0xF4 +#define ACON2 0xF8 +#define MC1 0xFC +#define MC2 0x100 +#define ISR 0x10C +#define PSR 0x110 +#define SSR 0x114 +#define PCI_ADP1 0x12C +#define PCI_ADP2 0x130 +#define PCI_ADP3 0x134 +#define PCI_ADP4 0x138 +#define LEVEL_REP 0x140 +#define FB_BUFFER1 0x144 +#define FB_BUFFER2 0x148 +#define TSL1 0x180 +#define TSL2 0x1C0 + +#define ME (1UL << 11) +#define LIMIT (1UL << 4) +#define PV (1UL << 3) + +/* PSR/ISR/IER */ +#define PPEF (1UL << 31) +#define PABO (1UL << 30) +#define IIC_S (1UL << 17) +#define IIC_E (1UL << 16) +#define A2_in (1UL << 15) +#define A2_out (1UL << 14) +#define A1_in (1UL << 13) +#define A1_out (1UL << 12) +#define AFOU (1UL << 11) +#define PIN3 (1UL << 6) +#define PIN2 (1UL << 5) +#define PIN1 (1UL << 4) +#define PIN0 (1UL << 3) +#define ECS (1UL << 2) +#define EC3S (1UL << 1) +#define EC0S (1UL << 0) + +/* SSR */ +#define PRQ (1UL << 31) +#define PMA (1UL << 30) +#define IIC_EA (1UL << 21) +#define IIC_EW (1UL << 20) +#define IIC_ER (1UL << 19) +#define IIC_EL (1UL << 18) +#define IIC_EF (1UL << 17) +#define AF2_in (1UL << 10) +#define AF2_out (1UL << 9) +#define AF1_in (1UL << 8) +#define AF1_out (1UL << 7) +#define EC5S (1UL << 3) +#define EC4S (1UL << 2) +#define EC2S (1UL << 1) +#define EC1S (1UL << 0) + +/* PCI_BT_A */ +#define BurstA1_in (1UL << 26) +#define ThreshA1_in (1UL << 24) +#define BurstA1_out (1UL << 18) +#define ThreshA1_out (1UL << 16) +#define BurstA2_in (1UL << 10) +#define ThreshA2_in (1UL << 8) +#define BurstA2_out (1UL << 2) +#define ThreshA2_out (1UL << 0) + +/* MC1 */ +#define MRST_N (1UL << 15) +#define EAP (1UL << 9) +#define EI2C (1UL << 8) +#define TR_E_A2_OUT (1UL << 3) +#define TR_E_A2_IN (1UL << 2) +#define TR_E_A1_OUT (1UL << 1) +#define TR_E_A1_IN (1UL << 0) + +/* MC2 */ +#define UPLD_IIC (1UL << 0) + +/* ACON1 */ +#define AUDIO_MODE (1UL << 29) +#define MAXLEVEL (1UL << 22) +#define A1_SWAP (1UL << 21) +#define A2_SWAP (1UL << 20) +#define WS0_CTRL (1UL << 18) +#define WS0_SYNC (1UL << 16) +#define WS1_CTRL (1UL << 14) +#define WS1_SYNC (1UL << 12) +#define WS2_CTRL (1UL << 10) +#define WS2_SYNC (1UL << 8) +#define WS3_CTRL (1UL << 6) +#define WS3_SYNC (1UL << 4) +#define WS4_CTRL (1UL << 2) +#define WS4_SYNC (1UL << 0) + +/* ACON2 */ +#define A1_CLKSRC (1UL << 27) +#define A2_CLKSRC (1UL << 22) +#define INVERT_BCLK1 (1UL << 21) +#define INVERT_BCLK2 (1UL << 20) +#define BCLK1_OEN (1UL << 19) +#define BCLK2_OEN (1UL << 18) + +/* IICSTA */ +#define IICCC (1UL << 8) +#define ABORT (1UL << 7) +#define SPERR (1UL << 6) +#define APERR (1UL << 5) +#define DTERR (1UL << 4) +#define DRERR (1UL << 3) +#define AL (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +/* IICTFR */ +#define BYTE2 (1UL << 24) +#define BYTE1 (1UL << 16) +#define BYTE0 (1UL << 8) +#define ATRR2 (1UL << 6) +#define ATRR1 (1UL << 4) +#define ATRR0 (1UL << 2) +#define ERR (1UL << 1) +#define BUSY (1UL << 0) + +#define START 3 +#define CONT 2 +#define STOP 1 +#define NOP 0 diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c new file mode 100644 index 0000000..f954b0b --- /dev/null +++ b/sound/pci/aw2/aw2-alsa.c @@ -0,0 +1,753 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/control.h> +#include <linux/interrupt.h> +#include <asm/io.h> +#include "SAA7146.h" + +#include <linux/delay.h> + +#include "aw2-saa7146.h" + +MODULE_LICENSE("GPL"); +/********************************* + * DEFINES + ********************************/ +#define PCI_VENDOR_ID_SAA7146 0x1131 +#define PCI_DEVICE_ID_SAA7146 0x7146 + +#define CTL_ROUTE_ANALOG 0 +#define CTL_ROUTE_DIGITAL 1 + +/********************************* + * TYPEDEFS + ********************************/ + /* hardware definition */ +static struct snd_pcm_hardware snd_aw2_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 4, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +static struct snd_pcm_hardware snd_aw2_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, +}; + +struct aw2_pcm_device_t { + struct snd_pcm *pcm; + unsigned int stream_number; + struct aw2_t *chip; +}; + +struct aw2_t { + struct snd_aw2_saa7146_t saa7146; + + struct snd_card *card; + + struct aw2_pcm_device_t device_playback[NB_STREAM_PLAYBACK]; + struct aw2_pcm_device_t device_capture[NB_STREAM_CAPTURE]; +}; + +/********************************* + * FUNCTION DECLARATIONS + ********************************/ +static int __init alsa_card_aw2_init(void); +static void __exit alsa_card_aw2_exit(void); +static int snd_aw2_dev_free(struct snd_device *device); +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2_t **rchip); +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id); +static void __devexit snd_aw2_remove(struct pci_dev *pci); +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params); +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream); +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd); +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream); +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream); +static int __devinit snd_aw2_new_pcm(struct aw2_t *chip); + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol); + +/********************************* + * VARIABLES + ********************************/ +/* module parameters (see "Module Parameters") */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +static struct pci_device_id snd_aw2_ids[] = { + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + 0, 0, 0}, + {0} +}; + +MODULE_DEVICE_TABLE(pci, snd_aw2_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = "My Own Chip", + .id_table = snd_aw2_ids, + .probe = snd_aw2_probe, + .remove = __devexit_p(snd_aw2_remove), +}; + +/* operators for playback PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_playback_ops = { + .open = snd_aw2_pcm_playback_open, + .close = snd_aw2_pcm_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_playback, + .trigger = snd_aw2_pcm_trigger_playback, + .pointer = snd_aw2_pcm_pointer_playback, +}; + +/* operators for capture PCM alsa interface */ +static struct snd_pcm_ops snd_aw2_capture_ops = { + .open = snd_aw2_pcm_capture_open, + .close = snd_aw2_pcm_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aw2_pcm_hw_params, + .hw_free = snd_aw2_pcm_hw_free, + .prepare = snd_aw2_pcm_prepare_capture, + .trigger = snd_aw2_pcm_trigger_capture, + .pointer = snd_aw2_pcm_pointer_capture, +}; + +static struct snd_kcontrol_new aw2_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Capture Route", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = 0xffff, + .info = snd_aw2_control_switch_capture_info, + .get = snd_aw2_control_switch_capture_get, + .put = snd_aw2_control_switch_capture_put +}; + +/********************************* + * FUNCTION IMPLEMENTATIONS + ********************************/ + +/* initialization of the module */ +static int __init alsa_card_aw2_init(void) +{ + printk(KERN_DEBUG "Load aw2 module\n"); + return pci_register_driver(&driver); +} + +/* clean up the module */ +static void __exit alsa_card_aw2_exit(void) +{ + printk(KERN_DEBUG "Unload aw2 module\n"); + pci_unregister_driver(&driver); +} + +module_init(alsa_card_aw2_init); +module_exit(alsa_card_aw2_exit); + +/* component-destructor + * (see "Management of Cards and Components") + */ +static int snd_aw2_dev_free(struct snd_device *device) +{ + struct aw2_t *chip = device->device_data; + return snd_aw2_saa7146_free(&chip->saa7146); +} + +/* chip-specific constructor + * (see "Management of Cards and Components") + */ +static int __devinit snd_aw2_create(struct snd_card *card, + struct pci_dev *pci, struct aw2_t **rchip) +{ + struct aw2_t *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_aw2_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + pci_set_master(pci); + + /* check PCI availability (32bit DMA) */ + if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + printk(KERN_ERR "Impossible to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + chip = kcalloc(1, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->saa7146.pci = pci; + chip->saa7146.irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "Audiowerk2"); + if (err < 0) { + kfree(chip); + return err; + } + chip->saa7146.iobase_phys = pci_resource_start(pci, 0); + chip->saa7146.iobase_virt = + ioremap_nocache(chip->saa7146.iobase_phys, + pci_resource_len(pci, 0)); + + if (request_irq(pci->irq, snd_aw2_saa7146_interrupt, + IRQF_SHARED, "Audiowerk2", (void *)chip)) { + printk(KERN_ERR "Cannot grab irq %d\n", pci->irq); + snd_aw2_saa7146_free(&chip->saa7146); + return -EBUSY; + } + chip->saa7146.irq = pci->irq; + + /* (2) initialization of the chip hardware */ + snd_aw2_saa7146_setup(&chip->saa7146); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_aw2_saa7146_free(&chip->saa7146); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + + /* Activate this portion to dump all the saa7146 registers : */ +#if 0 + snd_aw2_saa7146_dump_registers(&chip->saa7146); +#endif + + printk(KERN_INFO + "Audiowerk 2 sound card (saa7146 chipset) detected and " + "managed\n"); + return 0; +} + +/* constructor -- see "Constructor" sub-section */ +static int __devinit snd_aw2_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct aw2_t *chip; + int err; + + /* (1) Continue if device is not enabled, else inc dev */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) Create card instance */ + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + /* (3) Create main component */ + err = snd_aw2_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* initialize semaphore */ + init_MUTEX(&(chip->saa7146.sem)); + /* init spinlock */ + spin_lock_init(&chip->saa7146.reg_lock); + /* initialize wait queue */ + init_waitqueue_head(&chip->saa7146.wait_iic); + /* (4) Define driver ID and name string */ + strcpy(card->driver, "aw2"); + strcpy(card->shortname, "Audiowerk2"); + + sprintf(card->longname, "%s with SAA7146 at 0x%lx irq %i", + card->shortname, chip->saa7146.port, chip->saa7146.irq); + + /* (5) Create other components */ + snd_aw2_new_pcm(chip); + + /* (6) Register card instance */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) Set PCI driver data */ + pci_set_drvdata(pci, card); + + dev++; + return 0; +} + +/* destructor -- see "Destructor" sub-section */ +static void __devexit snd_aw2_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* open callback */ +static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + printk(KERN_DEBUG "Playback_open \n"); + runtime->hw = snd_aw2_playback_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_playback_close(struct snd_pcm_substream *substream) +{ + return 0; + +} + +static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + printk(KERN_DEBUG "Capture_open \n"); + runtime->hw = snd_aw2_capture_hw; + return 0; +} + +/* close callback */ +static int snd_aw2_pcm_capture_close(struct snd_pcm_substream *substream) +{ + /* TODO: something to do ? */ + return 0; +} + + /* hw_params callback */ +static int snd_aw2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_aw2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare callback for playback */ +static int snd_aw2_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + down(&chip->saa7146.sem); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_playback_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + up(&chip->saa7146.sem); + + return 0; +} + +/* prepare callback for capture */ +static int snd_aw2_pcm_prepare_capture(struct snd_pcm_substream *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long period_size, buffer_size; + + down(&chip->saa7146.sem); + + period_size = snd_pcm_lib_period_bytes(substream); + buffer_size = snd_pcm_lib_buffer_bytes(substream); + + snd_aw2_saa7146_pcm_init_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_addr, period_size, + buffer_size); + + /* Define Interrupt callback */ + snd_aw2_saa7146_define_it_capture_callback(pcm_device->stream_number, + (snd_aw2_saa7146_it_cb) + snd_pcm_period_elapsed, + (void *)substream); + + up(&chip->saa7146.sem); + + return 0; +} + +/* playback trigger callback */ +static int snd_aw2_pcm_trigger_playback(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + spin_lock(&chip->saa7146.reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_playback(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->saa7146.reg_lock); + return status; +} + +/* capture trigger callback */ +static int snd_aw2_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + int status = 0; + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + spin_lock(&chip->saa7146.reg_lock); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_aw2_saa7146_pcm_trigger_start_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_aw2_saa7146_pcm_trigger_stop_capture(&chip->saa7146, + pcm_device-> + stream_number); + break; + default: + status = -EINVAL; + } + spin_unlock(&chip->saa7146.reg_lock); + return status; +} + +/* playback pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_playback(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_pointer_playback(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* capture pointer callback */ +static snd_pcm_uframes_t snd_aw2_pcm_pointer_capture(struct snd_pcm_substream + *substream) +{ + struct aw2_pcm_device_t *pcm_device = snd_pcm_substream_chip(substream); + struct aw2_t *chip = pcm_device->chip; + unsigned int current_ptr; + + /* get the current hardware pointer */ + struct snd_pcm_runtime *runtime = substream->runtime; + current_ptr = + snd_aw2_saa7146_get_hw_pointer_capture(&chip->saa7146, + pcm_device->stream_number, + runtime->dma_area, + runtime->buffer_size); + + return bytes_to_frames(substream->runtime, current_ptr); +} + +/* create a pcm device */ +static int __devinit snd_aw2_new_pcm(struct aw2_t *chip) +{ + struct snd_pcm *pcm_playback_ana; + struct snd_pcm *pcm_playback_num; + struct snd_pcm *pcm_capture; + int err = 0; + + /* Create new Alsa PCM device */ + + err = + snd_pcm_new(chip->card, "Audiowerk2 analog playback", 0, 1, 0, + &pcm_playback_ana); + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_playback[NUM_STREAM_PLAYBACK_ANA]; + + /* Set PCM device name */ + strcpy(pcm_playback_ana->name, "Analog playback"); + /* Associate private data to PCM device */ + pcm_playback_ana->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_ana, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_ana; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_playback_ana, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + err = + snd_pcm_new(chip->card, "Audiowerk2 digital playback", 1, 1, 0, + &pcm_playback_num); + + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_playback[NUM_STREAM_PLAYBACK_DIG]; + + /* Set PCM device name */ + strcpy(pcm_playback_num->name, "Digital playback"); + /* Associate private data to PCM device */ + pcm_playback_num->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_playback_num, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aw2_playback_ops); + /* store PCM device */ + pcm_device->pcm = pcm_playback_num; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_PLAYBACK_DIG; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_playback_num, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + err = snd_pcm_new(chip->card, "Audiowerk2 capture", 2, 0, 1, + &pcm_capture); + + if (err >= 0) { + /* Creation ok */ + struct aw2_pcm_device_t *pcm_device = + &chip->device_capture[NUM_STREAM_CAPTURE_ANA]; + + /* Set PCM device name */ + strcpy(pcm_capture->name, "Capture"); + /* Associate private data to PCM device */ + pcm_capture->private_data = pcm_device; + /* set operators of PCM device */ + snd_pcm_set_ops(pcm_capture, SNDRV_PCM_STREAM_CAPTURE, + &snd_aw2_capture_ops); + /* store PCM device */ + pcm_device->pcm = pcm_capture; + /* give base chip pointer to our internal pcm device + structure */ + pcm_device->chip = chip; + /* Give stream number to PCM device */ + pcm_device->stream_number = NUM_STREAM_CAPTURE_ANA; + + /* pre-allocation of buffers */ + /* Preallocate continuous pages. */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm_capture, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data + (chip->saa7146.pci), + 64 * 1024, 64 * 1024); + if (err) { + printk(KERN_ERR + "snd_pcm_lib_preallocate_pages_for_all error " + "(0x%X)\n", err); + } + } + + /* Create control */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&aw2_control, chip)); + if (err < 0) { + printk(KERN_ERR "snd_ctl_add error (0x%X)\n", err); + } + return 0; +} + +static int snd_aw2_control_switch_capture_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[2] = { + "Analog", "Digital" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) { + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + } + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_aw2_control_switch_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2_t *chip = snd_kcontrol_chip(kcontrol); + if (snd_aw2_saa7146_is_using_digital_input(&chip->saa7146)) { + ucontrol->value.enumerated.item[0] = CTL_ROUTE_DIGITAL; + } else { + ucontrol->value.enumerated.item[0] = CTL_ROUTE_ANALOG; + } + return 0; +} + +static int snd_aw2_control_switch_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value + *ucontrol) +{ + struct aw2_t *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + int is_disgital = + snd_aw2_saa7146_is_using_digital_input(&chip->saa7146); + + if (((ucontrol->value.integer.value[0] == CTL_ROUTE_DIGITAL) + && !is_disgital) + || ((ucontrol->value.integer.value[0] == CTL_ROUTE_ANALOG) + && is_disgital)) { + snd_aw2_saa7146_use_digital_input(&chip->saa7146, !is_disgital); + changed = 1; + } + return changed; +} diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c new file mode 100644 index 0000000..f3e180f --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.c @@ -0,0 +1,613 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define AW2_SAA7146_M + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include <linux/interrupt.h> +#include <asm/io.h> +#include "aw2_tsl.h" +#include "SAA7146.h" + +#include <linux/delay.h> + +#include <asm/system.h> + +#include "aw2-saa7146.h" + +#define WRITEREG(value, addr) do { writel((value), chip->iobase_virt + (addr));\ + mb(); } while (0) +#define READREG(addr) readl(chip->iobase_virt + (addr)) + +static struct snd_aw2_saa7146_cb_param + arr_substream_it_playback_cb[NB_STREAM_PLAYBACK]; +static struct snd_aw2_saa7146_cb_param + arr_substream_it_capture_cb[NB_STREAM_CAPTURE]; + +static int snd_aw2_saa7146_i2c_write(struct snd_aw2_saa7146_t *chip, + unsigned char address, const void *buf, + size_t count); +static int snd_aw2_saa7146_get_limit(int size); + +void snd_aw2_saa7146_dump_registers(struct snd_aw2_saa7146_t *chip) +{ + int i = 0; + for (i = 0; i <= 0x148; i += 4) { + printk(KERN_DEBUG "###DUMP 0x%03x: 0x%08x\n", i, READREG(i)); + } +} + +/* chip-specific destructor + * (see "PCI Resource Managements") + */ +int snd_aw2_saa7146_free(struct snd_aw2_saa7146_t *chip) +{ + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* release the irq */ + if (chip->irq >= 0) { + free_irq(chip->irq, (void *)chip); + } + /* release the i/o ports & memory */ + if (chip->iobase_virt) { + iounmap(chip->iobase_virt); + } + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; +} + +void snd_aw2_saa7146_setup(struct snd_aw2_saa7146_t *chip) +{ + /* set PCI burst/threshold + + Burst length definition + VALUE BURST LENGTH + 000 1 Dword + 001 2 Dwords + 010 4 Dwords + 011 8 Dwords + 100 16 Dwords + 101 32 Dwords + 110 64 Dwords + 111 128 Dwords + + Threshold definition + VALUE WRITE MODE READ MODE + 00 1 Dword of valid data 1 empty Dword + 01 4 Dwords of valid data 4 empty Dwords + 10 8 Dwords of valid data 8 empty Dwords + 11 16 Dwords of valid data 16 empty Dwords */ + + unsigned int acon2; + unsigned int acon1 = 0; + int i; + + /* disable all irqs */ + WRITEREG(0, IER); + + /* reset saa7146 */ + WRITEREG((MRST_N << 16), MC1); + + /* enable audio interface */ +#ifdef __BIG_ENDIAN + acon1 |= A1_SWAP; + acon1 |= A2_SWAP; +#endif + /* WS0_CTRL, WS0_SYNC: input TSL1, I2S */ + + /* At initialization WS1 and WS2 are disbaled (configured as input */ + acon1 |= 0 * WS1_CTRL; + acon1 |= 0 * WS2_CTRL; + + /* WS4 is not used. So it must not restart A2. + This is why it is configured as output (force to low) */ + acon1 |= 3 * WS4_CTRL; + + /* WS3_CTRL, WS3_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS3_CTRL; + + /* A1 and A2 are active and asynchronous */ + acon1 |= 3 * AUDIO_MODE; + WRITEREG(acon1, ACON1); + + /* The following comes from original windows driver. + It is needed to have a correct behavior of input and output + simultenously, but I don't know why ! */ + WRITEREG(3 * (BurstA1_in) + 3 * (ThreshA1_in) + + 3 * (BurstA1_out) + 3 * (ThreshA1_out) + + 3 * (BurstA2_out) + 3 * (ThreshA2_out), PCI_BT_A); + + /* enable audio port pins */ + WRITEREG((EAP << 16) | EAP, MC1); + + /* enable I2C */ + WRITEREG((EI2C << 16) | EI2C, MC1); + /* enable interrupts */ + WRITEREG(A1_out | A2_out | A1_in | IIC_S | IIC_E, IER); + + /* audio configuration */ + acon2 = A2_CLKSRC | BCLK1_OEN; + WRITEREG(acon2, ACON2); + + /* By default use analog input */ + snd_aw2_saa7146_use_digital_input(chip, 0); + + /* TSL setup */ + for (i = 0; i < 8; ++i) { + WRITEREG(tsl1[i], TSL1 + (i * 4)); + WRITEREG(tsl2[i], TSL2 + (i * 4)); + } + +} + +void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA2_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA2_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA2_out); + + } else if (stream_number == 1) { + WRITEREG(dw_page, PageA1_out); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_out); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_out); + } else { + printk(KERN_ERR + "snd_aw2_saa7146_pcm_init_playback: Substream number is " + "not 0 or 1 -> not managed\n"); + } +} + +void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146_t *chip, + int stream_number, unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size) +{ + unsigned long dw_page, dw_limit; + + /* Configure DMA for substream + Configuration informations: ALSA has allocated continuous memory + pages. So we don't need to use MMU of saa7146. + */ + + /* No MMU -> nothing to do with PageA1, we only configure the limit of + PageAx_out register */ + /* Disable MMU */ + dw_page = (0L << 11); + + /* Configure Limit for DMA access. + The limit register defines an address limit, which generates + an interrupt if passed by the actual PCI address pointer. + '0001' means an interrupt will be generated if the lower + 6 bits (64 bytes) of the PCI address are zero. '0010' + defines a limit of 128 bytes, '0011' one of 256 bytes, and + so on up to 1 Mbyte defined by '1111'. This interrupt range + can be calculated as follows: + Range = 2^(5 + Limit) bytes. + */ + dw_limit = snd_aw2_saa7146_get_limit(period_size); + dw_page |= (dw_limit << 4); + + if (stream_number == 0) { + WRITEREG(dw_page, PageA1_in); + + /* Base address for DMA transfert. */ + /* This address has been reserved by ALSA. */ + /* This is a physical address */ + WRITEREG(dma_addr, BaseA1_in); + + /* Define upper limit for DMA access */ + WRITEREG(dma_addr + buffer_size, ProtA1_in); + } else { + printk(KERN_ERR + "snd_aw2_saa7146_pcm_init_capture: Substream number is " + "not 0 -> not managed\n"); + } +} + +void snd_aw2_saa7146_define_it_playback_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_PLAYBACK) { + arr_substream_it_playback_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_playback_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_define_it_capture_callback(unsigned int stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param) +{ + if (stream_number < NB_STREAM_CAPTURE) { + arr_substream_it_capture_cb[stream_number].p_it_callback = + (snd_aw2_saa7146_it_cb) p_it_callback; + arr_substream_it_capture_cb[stream_number].p_callback_param = + (void *)p_callback_param; + } +} + +void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + unsigned int acon1 = 0; + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + acon1 = READREG(ACON1); + if (stream_number == 0) { + WRITEREG((TR_E_A2_OUT << 16) | TR_E_A2_OUT, MC1); + + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 |= 2 * WS2_CTRL; + WRITEREG(acon1, ACON1); + + } else if (stream_number == 1) { + WRITEREG((TR_E_A1_OUT << 16) | TR_E_A1_OUT, MC1); + + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 |= 1 * WS1_CTRL; + WRITEREG(acon1, ACON1); + } else { + } + +} + +void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + unsigned int acon1 = 0; + acon1 = READREG(ACON1); + if (stream_number == 0) { + /* WS2_CTRL, WS2_SYNC: output TSL2, I2S */ + acon1 &= ~(3 * WS2_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A2_OUT << 16), MC1); + } else if (stream_number == 1) { + /* WS1_CTRL, WS1_SYNC: output TSL1, I2S */ + acon1 &= ~(3 * WS1_CTRL); + WRITEREG(acon1, ACON1); + + WRITEREG((TR_E_A1_OUT << 16), MC1); + } else { + } +} + +void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + /* In aw8 driver, dma transfert is always active. It is + started and stopped in a larger "space" */ + if (stream_number == 0) { + WRITEREG((TR_E_A1_IN << 16) | TR_E_A1_IN, MC1); + } else { + } +} + +void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146_t *chip, + int stream_number) +{ + if (stream_number == 0) { + WRITEREG((TR_E_A1_IN << 16), MC1); + } else { + } + +} + +irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id) +{ + unsigned int isr; + unsigned int iicsta; + struct snd_aw2_saa7146_t *chip = dev_id; + + isr = READREG(ISR); + if (!isr) + return IRQ_NONE; + + WRITEREG(isr, ISR); + + if (isr & (IIC_S | IIC_E)) { + iicsta = READREG(IICSTA); + WRITEREG(0x100, IICSTA); + wake_up(&chip->wait_iic); + } else if (isr) { + } + + if (isr & A1_out) { + if (arr_substream_it_playback_cb[1].p_it_callback != NULL) { + arr_substream_it_playback_cb[1]. + p_it_callback(arr_substream_it_playback_cb[1]. + p_callback_param); + } + } + if (isr & A2_out) { + if (arr_substream_it_playback_cb[0].p_it_callback != NULL) { + arr_substream_it_playback_cb[0]. + p_it_callback(arr_substream_it_playback_cb[0]. + p_callback_param); + } + + } + if (isr & A1_in) { + if (arr_substream_it_capture_cb[0].p_it_callback != NULL) { + arr_substream_it_capture_cb[0]. + p_it_callback(arr_substream_it_capture_cb[0]. + p_callback_param); + } + } + return IRQ_HANDLED; +} + +unsigned int snd_aw2_saa7146_get_hw_pointer_playback(struct snd_aw2_saa7146_t + *chip, int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + long pci_adp = 0; + size_t ptr = 0; + + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP3); + ptr = pci_adp - (long)start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } + if (stream_number == 1) { + pci_adp = READREG(PCI_ADP1); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } else { + } + return ptr; +} + +unsigned int snd_aw2_saa7146_get_hw_pointer_capture(struct snd_aw2_saa7146_t + *chip, int stream_number, + unsigned char *start_addr, + unsigned int buffer_size) +{ + size_t pci_adp = 0; + size_t ptr = 0; + if (stream_number == 0) { + pci_adp = READREG(PCI_ADP2); + ptr = pci_adp - (size_t) start_addr; + + if (ptr == buffer_size) { + ptr = 0; + } + } else { + } + return ptr; +} + +int snd_aw2_saa7146_set_sample_rate(struct snd_aw2_saa7146_t *chip, int srate) +{ + unsigned int S; + unsigned char DB[4]; + unsigned int iicsta = 0; + int err = 0; + /* check srate range */ + if ((srate < 30000) || (srate > 50000)) + return -EINVAL; + + /* PLL output is 256*fs, reference 1kHz */ + S = srate * 256 / 1000; + + /* LSB is at S2 */ + S <<= 2; + + DB[0] = (S << 1) & 0xff; + DB[1] = (S >> 7) & 0xff; + DB[2] = ((S >> 15) & 0xff) | 0x18; + DB[3] = 0; + + if (chip->iic) + return -EBUSY; + chip->iic = 1; + iicsta = READREG(IICSTA); + up(&chip->sem); + err = snd_aw2_saa7146_i2c_write(chip, 0xc4, DB, sizeof(DB)); + down(&chip->sem); + chip->iic = 0; + return err; + +} + +void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146_t *chip, + int use_digital) +{ + /* FIXME: switch between analog and digital input does not always work. + It can produce a kind of white noise. It seams that received data + are inverted sometime (endian inversion). Why ? I don't know, maybe + a problem of synchronization... However for the time being I have + not found the problem. Workaround: switch again (and again) between + digital and analog input until it works. */ + if (use_digital) { + WRITEREG(0x40, GPIO_CTRL); + } else { + WRITEREG(0x50, GPIO_CTRL); + } +} + +int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146_t *chip) +{ + unsigned int reg_val = READREG(GPIO_CTRL); + if ((reg_val & 0xFF) == 0x40) { + return 1; + } else { + return 0; + } +} + +static int snd_aw2_saa7146_i2c_write(struct snd_aw2_saa7146_t *chip, + unsigned char address, const void *buf, + size_t count) +{ + unsigned int iicsta = 0; + unsigned int iictfr = 0; + int i, b; + + /* We only keep low significant bits i.e. without clock setting */ + iicsta = (READREG(IICSTA) & 0xFF); + + /* Clock setting */ + iicsta = 1 * IICCC; + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + if ((iicsta & BUSY) || (iicsta & ERR)) { /* error or busy */ + WRITEREG(iicsta | ABORT, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + /* clear error bits (2x) */ + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + WRITEREG(iicsta, IICSTA); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + } + + iictfr = address * BYTE2; + iictfr |= START * ATRR2; + + for (i = 0; i <= count; ++i) { + b = i % 3; + if ((b == 0) || (i == count)) { + wait_queue_t wait; + init_waitqueue_entry(&wait, current); + + add_wait_queue(&chip->wait_iic, &wait); + set_current_state(TASK_INTERRUPTIBLE); + + WRITEREG(iictfr, IICTFR); + WRITEREG((UPLD_IIC << 16) | UPLD_IIC, MC2); + + schedule(); + set_current_state(TASK_RUNNING); + remove_wait_queue(&chip->wait_iic, &wait); + + if (signal_pending(current)) { + return -ERESTARTSYS; + } + + iicsta = READREG(IICSTA); + if (iicsta & ERR) { + return -EIO; + } + if (i == count) + break; + iictfr = 0; + } + switch (b) { + case 0: + iictfr |= BYTE2 * ((char *)buf)[i]; + iictfr |= ATRR2 * ((count - i == 1) ? STOP : CONT); + break; + case 1: + iictfr |= BYTE1 * ((char *)buf)[i]; + iictfr |= ATRR1 * ((count - i == 1) ? STOP : CONT); + break; + case 2: + iictfr |= BYTE0 * ((char *)buf)[i]; + iictfr |= ATRR0 * ((count - i == 1) ? STOP : CONT); + break; + } + } + return 0; +} + +static int snd_aw2_saa7146_get_limit(int size) +{ + int limitsize = 32; + int limit = 0; + while (limitsize < size) { + limitsize *= 2; + limit++; + } + return limit; +} diff --git a/sound/pci/aw2/aw2-saa7146.h b/sound/pci/aw2/aw2-saa7146.h new file mode 100644 index 0000000..93e4883 --- /dev/null +++ b/sound/pci/aw2/aw2-saa7146.h @@ -0,0 +1,125 @@ +/***************************************************************************** + * + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#ifndef AW2_SAA7146 +#define AW2_SAA7146 + +#ifdef AW2_SAA7146_M +#define PUBLIC +#else +#define PUBLIC extern +#endif + +#define NB_STREAM_PLAYBACK 2 +#define NB_STREAM_CAPTURE 1 + +#define NUM_STREAM_PLAYBACK_ANA 0 +#define NUM_STREAM_PLAYBACK_DIG 1 + +#define NUM_STREAM_CAPTURE_ANA 0 + +typedef void (*snd_aw2_saa7146_it_cb) (void *); + +struct snd_aw2_saa7146_cb_param { + snd_aw2_saa7146_it_cb p_it_callback; + void *p_callback_param; +}; + +/* definition of the chip-specific record */ + +struct snd_aw2_saa7146_t { + struct pci_dev *pci; + + unsigned long iobase_phys; + void __iomem *iobase_virt; + unsigned long port; + int irq; + + struct semaphore sem; + spinlock_t reg_lock; + int iic; + wait_queue_head_t wait_iic; +}; + +PUBLIC void snd_aw2_saa7146_dump_registers(struct snd_aw2_saa7146_t *chip); + +PUBLIC int snd_aw2_saa7146_free(struct snd_aw2_saa7146_t *chip); +PUBLIC void snd_aw2_saa7146_setup(struct snd_aw2_saa7146_t *chip); +PUBLIC void snd_aw2_saa7146_pcm_init_playback(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +PUBLIC void snd_aw2_saa7146_pcm_init_capture(struct snd_aw2_saa7146_t *chip, + int stream_number, + unsigned long dma_addr, + unsigned long period_size, + unsigned long buffer_size); +PUBLIC void snd_aw2_saa7146_define_it_playback_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +PUBLIC void snd_aw2_saa7146_define_it_capture_callback(unsigned int + stream_number, + snd_aw2_saa7146_it_cb + p_it_callback, + void *p_callback_param); +PUBLIC void snd_aw2_saa7146_pcm_trigger_start_capture(struct snd_aw2_saa7146_t + *chip, int stream_number); +PUBLIC void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146_t + *chip, int stream_number); + +PUBLIC void snd_aw2_saa7146_pcm_trigger_start_playback(struct snd_aw2_saa7146_t + *chip, + int stream_number); +PUBLIC void snd_aw2_saa7146_pcm_trigger_stop_playback(struct snd_aw2_saa7146_t + *chip, int stream_number); + +PUBLIC irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id); +PUBLIC unsigned int snd_aw2_saa7146_get_hw_pointer_playback(struct + snd_aw2_saa7146_t + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +PUBLIC unsigned int snd_aw2_saa7146_get_hw_pointer_capture(struct + snd_aw2_saa7146_t + *chip, + int stream_number, + unsigned char + *start_addr, + unsigned int + buffer_size); +PUBLIC int snd_aw2_saa7146_set_sample_rate(struct snd_aw2_saa7146_t *chip, + int srate); + +PUBLIC void snd_aw2_saa7146_use_digital_input(struct snd_aw2_saa7146_t *chip, + int use_digital); + +PUBLIC int snd_aw2_saa7146_is_using_digital_input(struct snd_aw2_saa7146_t + *chip); + +#endif diff --git a/sound/pci/aw2/aw2_tsl.h b/sound/pci/aw2/aw2_tsl.h new file mode 100644 index 0000000..af2edbf --- /dev/null +++ b/sound/pci/aw2/aw2_tsl.h @@ -0,0 +1,116 @@ +/***************************************************************************** + * + * Copyright 1998 Emagic Soft- und Hardware GmbH + * Copyright 2002 Martijn Sipkema + * Copyright (C) 2007 Cedric Bregardis <cedric.bregardis AT free.fr> and + * Jean-Christian Hassler <jhassler AT free.fr> + * + * This file is part of the Audiowerk2 ALSA driver + * + * The Audiowerk2 ALSA driver 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; version 2. + * + * The Audiowerk2 ALSA driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with the Audiowerk2 ALSA driver; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, + * USA. + * + *****************************************************************************/ + +#define TSL_WS0 (1UL << 31) +#define TSL_WS1 (1UL << 30) +#define TSL_WS2 (1UL << 29) +#define TSL_WS3 (1UL << 28) +#define TSL_WS4 (1UL << 27) +#define TSL_DIS_A1 (1UL << 24) +#define TSL_SDW_A1 (1UL << 23) +#define TSL_SIB_A1 (1UL << 22) +#define TSL_SF_A1 (1UL << 21) +#define TSL_LF_A1 (1UL << 20) +#define TSL_BSEL_A1 (1UL << 17) +#define TSL_DOD_A1 (1UL << 15) +#define TSL_LOW_A1 (1UL << 14) +#define TSL_DIS_A2 (1UL << 11) +#define TSL_SDW_A2 (1UL << 10) +#define TSL_SIB_A2 (1UL << 9) +#define TSL_SF_A2 (1UL << 8) +#define TSL_LF_A2 (1UL << 7) +#define TSL_BSEL_A2 (1UL << 4) +#define TSL_DOD_A2 (1UL << 2) +#define TSL_LOW_A2 (1UL << 1) +#define TSL_EOS (1UL << 0) + + /* Audiowerk8 hardware setup: */ + /* WS0, SD4, TSL1 - Analog/ digital in */ + /* WS1, SD0, TSL1 - Analog out #1, digital out */ + /* WS2, SD2, TSL1 - Analog out #2 */ + /* WS3, SD1, TSL2 - Analog out #3 */ + /* WS4, SD3, TSL2 - Analog out #4 */ + + /* Audiowerk8 timing: */ + /* Timeslot: | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | ... */ + + /* A1_INPUT: */ + /* SD4: <_ADC-L_>-------<_ADC-R_>-------< */ + /* WS0: _______________/---------------\_ */ + + /* A1_OUTPUT: */ + /* SD0: <_1-L___>-------<_1-R___>-------< */ + /* WS1: _______________/---------------\_ */ + /* SD2: >-------<_2-L___>-------<_2-R___> */ + /* WS2: -------\_______________/--------- */ + + /* A2_OUTPUT: */ + /* SD1: <_3-L___>-------<_3-R___>-------< */ + /* WS3: _______________/---------------\_ */ + /* SD3: >-------<_4-L___>-------<_4-R___> */ + /* WS4: -------\_______________/--------- */ + +#ifdef __BIG_ENDIAN + /* TODO: not yet implemented */ +#else /* */ + +static int tsl1[8] = { + 1 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_LF_A1, + + 1 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 3 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 0 * TSL_SDW_A1 | 2 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1, + + 1 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 1 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 1 * TSL_BSEL_A1 | + 0 * TSL_DIS_A1 | 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0, + + 0 * TSL_SDW_A1 | 0 * TSL_BSEL_A1 | 0 * TSL_DIS_A1 | + 0 * TSL_DOD_A1 | TSL_WS1 | TSL_WS0 | TSL_SF_A1 | TSL_EOS, +}; + +static int tsl2[8] = { + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_LF_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 3 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 2 * TSL_BSEL_A2 | 2 * TSL_DOD_A2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 1 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2, + 0 * TSL_SDW_A2 | 0 * TSL_BSEL_A2 | 2 * TSL_DOD_A2 | TSL_WS2 | TSL_EOS +}; + +#endif /* */ -- 1.5.2.5

2 1

[alsa-devel] [PATCH 1/1] Add __devinit macro to at73c213 sound driver probe functions
by Hans-Christian Egtvedt 18 Feb '08

18 Feb '08

This patch adds __devinit to the functions used when probing. Will also reduce the memory footprint a bit if CONFIG_HOTPLUG is not enabled. Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- sound/spi/at73c213.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index fee869b..83fa475 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -744,7 +744,7 @@ cleanup: /* * Device functions */ -static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { /* * Continuous clock output. @@ -774,7 +774,7 @@ static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) return 0; } -static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +static int __devinit snd_at73c213_chip_init(struct snd_at73c213 *chip) { int retval; unsigned char dac_ctrl = 0; @@ -939,7 +939,7 @@ out: return retval; } -static int snd_at73c213_probe(struct spi_device *spi) +static int __devinit snd_at73c213_probe(struct spi_device *spi) { struct snd_card *card; struct snd_at73c213 *chip; -- 1.5.2.5

2 1

[alsa-devel] [PATCH] Davinci ASoC support
by Vladimir Barinov 18 Feb '08

18 Feb '08

Add ASoC support for the TI Davinci SoC and the Davicni-EVM reference board. It includes: - ASoC Davinci DMA driver - ASoC Davinci I2S (Davinci McBSP module based) driver - ASoC Davinci-EVM reference board Signed-off-by: Vladimir Barinov <vbarinov(a)ru.mvista.com> Index: linux-2.6.25-rc1/sound/soc/davinci/Makefile =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/Makefile @@ -0,0 +1,11 @@ +# DAVINCI Platform Support +snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-i2s-objs := davinci-i2s.o + +obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o + +# DAVINCI Machine Support +snd-soc-evm-objs := davinci-evm.o + +obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-evm.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-evm.c @@ -0,0 +1,208 @@ +/* + * ASoC driver for TI DAVINCI EVM platform + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/dma.h> +#include <asm/arch/hardware.h> + +#include "../codecs/tlv320aic3x.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +#define EVM_CODEC_CLOCK 22579200 + +static int evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops evm_ops = { + .hw_params = evm_hw_params, +}; + +/* davinci-evm machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +/* davinci-evm machine audio_mapnections to the codec pins */ +static const char *audio_map[][3] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, + + {NULL, NULL, NULL}, +}; + +/* Logic for a aic3x as connected on a davinci-evm */ +static int evm_aic3x_init(struct snd_soc_codec *codec) +{ + int i; + + /* Add davinci-evm specific widgets */ + for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + + /* Set up davinci-evm specific audio path audio_map */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + /* not connected */ + snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); + snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); + snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + + /* always connected */ + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line In", 1); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +/* davinci-evm digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +/* davinci-evm audio machine driver */ +static struct snd_soc_machine snd_soc_machine_evm = { + .name = "DaVinci EVM", + .dai_link = &evm_dai, + .num_links = 1, +}; + +/* evm audio private data */ +static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_address = 0x1b, +}; + +/* evm audio subsystem */ +static struct snd_soc_device evm_snd_devdata = { + .machine = &snd_soc_machine_evm, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &evm_aic3x_setup, +}; + +static struct resource evm_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data evm_snd_data = { + .tx_dma_ch = DM644X_DMACH_MCBSP_TX, + .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +}; + +static struct platform_device *evm_snd_device; + +static int __init evm_init(void) +{ + int ret; + + evm_snd_device = platform_device_alloc("soc-audio", 0); + if (!evm_snd_device) + return -ENOMEM; + + platform_set_drvdata(evm_snd_device, &evm_snd_devdata); + evm_snd_devdata.dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = &evm_snd_data; + + ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, + ARRAY_SIZE(evm_snd_resources)); + if (ret) { + platform_device_put(evm_snd_device); + return ret; + } + + ret = platform_device_add(evm_snd_device); + if (ret) + platform_device_put(evm_snd_device); + + return ret; +} + +static void __exit evm_exit(void) +{ + platform_device_unregister(evm_snd_device); +} + +module_init(evm_init); +module_exit(evm_exit); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.c @@ -0,0 +1,407 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" + +#define DAVINCI_MCBSP_DRR_REG 0x00 +#define DAVINCI_MCBSP_DXR_REG 0x04 +#define DAVINCI_MCBSP_SPCR_REG 0x08 +#define DAVINCI_MCBSP_RCR_REG 0x0c +#define DAVINCI_MCBSP_XCR_REG 0x10 +#define DAVINCI_MCBSP_SRGR_REG 0x14 +#define DAVINCI_MCBSP_PCR_REG 0x24 + +#define DAVINCI_MCBSP_SPCR_RRST (1 << 0) +#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_XRST (1 << 16) +#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) +#define DAVINCI_MCBSP_SPCR_GRST (1 << 22) +#define DAVINCI_MCBSP_SPCR_FRST (1 << 23) +#define DAVINCI_MCBSP_SPCR_FREE (1 << 25) + +#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_XCR_XFIG (1 << 18) +#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) +#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) +#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) + +#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) +#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) +#define DAVINCI_MCBSP_PCR_FSRP (1 << 2) +#define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) +#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) +#define DAVINCI_MCBSP_PCR_FSRM (1 << 10) +#define DAVINCI_MCBSP_PCR_FSXM (1 << 11) + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +enum { + DAVINCI_MCBSP_WORD_8 = 0, + DAVINCI_MCBSP_WORD_12, + DAVINCI_MCBSP_WORD_16, + DAVINCI_MCBSP_WORD_20, + DAVINCI_MCBSP_WORD_24, + DAVINCI_MCBSP_WORD_32, +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { + .name = "I2S PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { + .name = "I2S PCM Stereo in", +}; + +struct davinci_mcbsp_dev { + void __iomem *base; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_mcbsp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Start frame sync */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST | + DAVINCI_MCBSP_SPCR_FRST, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} + +static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; + case SND_SOC_DAIFMT_CBM_CFM: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_IB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_interval *i = NULL; + int mcbsp_word_length; + u32 w; + + /* general line settings */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + DAVINCI_MCBSP_SPCR_RINTM(3) | + DAVINCI_MCBSP_SPCR_XINTM(3) | + DAVINCI_MCBSP_SPCR_FREE); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + mcbsp_word_length = DAVINCI_MCBSP_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + mcbsp_word_length = DAVINCI_MCBSP_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + mcbsp_word_length = DAVINCI_MCBSP_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + + return 0; +} + +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcbsp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcbsp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_i2s_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *pdata; + int ret; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "McBSP region already claimed\n"); + return -EBUSY; + } + + dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_region; + } + + cpu_dai->private_data = dev; + + dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_free_mem; + } + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + pdata = pdev->dev.platform_data; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + + return 0; + +err_free_mem: + kfree(dev); +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + return ret; +} + +static void davinci_i2s_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct resource *mem; + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +struct snd_soc_cpu_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params,}, + .dai_ops = { + .set_fmt = davinci_i2s_set_dai_fmt, + }, +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.h =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-i2s.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef _DAVINCI_I2S_H +#define _DAVINCI_I2S_H + +extern struct snd_soc_cpu_dai davinci_i2s_dai; + +#endif Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.c =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.c @@ -0,0 +1,389 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#include "davinci-pcm.h" + +#define DAVINCI_PCM_DEBUG 0 +#if DAVINCI_PCM_DEBUG +#define DPRINTK(x...) printk(KERN_DEBUG x) +#else +#define DPRINTK(x...) +#endif + +static struct snd_pcm_hardware davinci_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +struct davinci_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ + int master_lch; /* Master DMA channel */ + int slave_lch; /* Slave DMA channel */ + struct davinci_pcm_dma_params *params; /* DMA params */ +}; + +static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int lch = prtd->slave_lch; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + unsigned short src_bidx, dst_bidx; + unsigned int data_type; + unsigned int count; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " + "period_size=%x\n", lch, dma_pos, period_size); + + data_type = prtd->params->data_type; + count = period_size / data_type; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = prtd->params->dma_addr; + src_bidx = data_type; + dst_bidx = 0; + } else { + src = prtd->params->dma_addr; + dst = dma_pos; + src_bidx = 0; + dst_bidx = data_type; + } + + davinci_set_dma_src_params(lch, src, INCR, W8BIT); + davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); + davinci_set_dma_src_index(lch, src_bidx, 0); + davinci_set_dma_dest_index(lch, dst_bidx, 0); + davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +{ + struct snd_pcm_substream *substream = data; + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + + if (unlikely(ch_status != DMA_COMPLETE)) + return; + + if (snd_pcm_running(substream)) { + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } +} + +static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int tcc = TCC_ANY; + int ret; + + if (!dma_data) + return -ENODEV; + + prtd->params = dma_data; + + /* Request master DMA channel */ + ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + davinci_pcm_dma_irq, substream, + &prtd->master_lch, &tcc, EVENTQ_0); + if (ret) + return ret; + + /* Request slave DMA channel */ + ret = davinci_request_dma(PARAM_ANY, "Link", + NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); + if (ret) { + davinci_free_dma(prtd->master_lch); + return ret; + } + + /* Link slave DMA channel in loopback */ + davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + + return 0; +} + +static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_start_dma(prtd->master_lch); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_stop_dma(prtd->master_lch); + break; + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static int davinci_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct paramentry_descriptor temp; + + prtd->period = 0; + davinci_pcm_enqueue_dma(substream); + + /* Get slave channel dma params for master channel startup */ + davinci_get_dma_params(prtd->slave_lch, &temp); + davinci_set_dma_params(prtd->master_lch, &temp); + + return 0; +} + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + unsigned int offset; + dma_addr_t count; + dma_addr_t src, dst; + + spin_lock(&prtd->lock); + + davinci_dma_getposition(prtd->master_lch, &src, &dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = src - runtime->dma_addr; + else + count = dst - runtime->dma_addr;; + + spin_unlock(&prtd->lock); + + offset = bytes_to_frames(runtime, count); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + + prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + ret = davinci_pcm_dma_request(substream); + if (ret) { + printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); + kfree(prtd); + } + + return ret; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + + davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + + davinci_free_dma(prtd->slave_lch); + davinci_free_dma(prtd->master_lch); + + kfree(prtd); + + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .prepare = davinci_pcm_prepare, + .trigger = davinci_pcm_trigger, + .pointer = davinci_pcm_pointer, + .mmap = davinci_pcm_mmap, +}; + +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = davinci_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void davinci_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform davinci_soc_platform = { + .name = "davinci-audio", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, + .pcm_free = davinci_pcm_free, +}; +EXPORT_SYMBOL_GPL(davinci_soc_platform); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); +MODULE_LICENSE("GPL"); Index: linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.h =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/davinci-pcm.h @@ -0,0 +1,29 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov(a)ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef _DAVINCI_PCM_H +#define _DAVINCI_PCM_H + +struct davinci_pcm_dma_params { + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ +}; + +struct evm_snd_platform_data { + int tx_dma_ch; + int rx_dma_ch; +}; + +extern struct snd_soc_platform davinci_soc_platform; + +#endif Index: linux-2.6.25-rc1/sound/soc/Kconfig =================================================================== --- linux-2.6.25-rc1.orig/sound/soc/Kconfig +++ linux-2.6.25-rc1/sound/soc/Kconfig @@ -29,6 +29,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/davinci/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" Index: linux-2.6.25-rc1/sound/soc/Makefile =================================================================== --- linux-2.6.25-rc1.orig/sound/soc/Makefile +++ linux-2.6.25-rc1/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ Index: linux-2.6.25-rc1/sound/soc/davinci/Kconfig =================================================================== --- /dev/null +++ linux-2.6.25-rc1/sound/soc/davinci/Kconfig @@ -0,0 +1,19 @@ +config SND_DAVINCI_SOC + tristate "SoC Audio for the TI DAVINCI chip" + depends on ARCH_DAVINCI && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the DAVINCI AC97 or I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_DAVINCI_SOC_I2S + tristate + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci EVM platform.

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