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openwrt/target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch

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--- a/arch/arm/mach-ep93xx/include/mach/hardware.h
+++ b/arch/arm/mach-ep93xx/include/mach/hardware.h
@@ -5,6 +5,7 @@
#define __ASM_ARCH_HARDWARE_H
#include "ep93xx-regs.h"
+#include "regs_ac97.h"
#define pcibios_assign_all_busses() 0
#include "regs_raster.h"
--- /dev/null
+++ b/arch/arm/mach-ep93xx/include/mach/regs_ac97.h
@@ -0,0 +1,180 @@
+/*=============================================================================
+ * FILE: regs_ac97.h
+ *
+ * DESCRIPTION: Ac'97 Register Definition
+ *
+ * Copyright Cirrus Logic, 2001-2003
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *=============================================================================
+ */
+#ifndef _REGS_AC97_H_
+#define _REGS_AC97_H_
+
+//-----------------------------------------------------------------------------
+// Bit definitionses
+//-----------------------------------------------------------------------------
+#define AC97ISR_RIS 8
+#define AC97ISR_TIS 4
+#define AC97ISR_RTIS 2
+#define AC97ISR_TCIS 1
+
+#define AC97RGIS_SLOT1TXCOMPLETE 0x01
+#define AC97RGIS_SLOT2RXVALID 0x02
+#define AC97RGIS_GPIOTXCOMPLETE 0x04
+#define AC97RGIS_GPIOINTRX 0x08
+#define AC97RGIS_RWIS 0x10
+#define AC97RGIS_CODECREADY 0x20
+#define AC97RGIS_SLOT2TXCOMPLETE 0x40
+
+#define AC97SR_RXFE 0x0001
+#define AC97SR_TXFE 0x0002
+#define AC97SR_RXFF 0x0004
+#define AC97SR_TXFF 0x0008
+#define AC97SR_TXBUSY 0x0010
+#define AC97SR_RXOE 0x0020
+#define AC97SR_TXUE 0x0040
+
+#define AC97GSR_IFE 0x1
+#define AC97GSR_LOOP 0x2
+#define AC97GSR_OVERRIDECODECREADY 0x4
+
+#define AC97RESET_TIMEDRESET 0x1
+#define AC97RESET_FORCEDRESET 0x2
+#define AC97RESET_EFORCER 0x4
+
+#define AC97RXCR_REN 0x1
+
+#define AC97TXCR_TEN 0x1
+
+
+//****************************************************************************
+//
+// The Ac97 Codec registers, accessable through the Ac-link.
+// These are not controller registers and are not memory mapped.
+// Includes registers specific to CS4202 (Beavis).
+//
+//****************************************************************************
+#define AC97_REG_OFFSET_MASK 0x0000007E
+
+#define AC97_00_RESET 0x00000000
+#define AC97_02_MASTER_VOL 0x00000002
+#define AC97_04_HEADPHONE_VOL 0x00000004
+#define AC97_06_MONO_VOL 0x00000006
+#define AC97_08_TONE 0x00000008
+#define AC97_0A_PC_BEEP_VOL 0x0000000A
+#define AC97_0C_PHONE_VOL 0x0000000C
+#define AC97_0E_MIC_VOL 0x0000000E
+#define AC97_10_LINE_IN_VOL 0x00000010
+#define AC97_12_CD_VOL 0x00000012
+#define AC97_14_VIDEO_VOL 0x00000014
+#define AC97_16_AUX_VOL 0x00000016
+#define AC97_18_PCM_OUT_VOL 0x00000018
+#define AC97_1A_RECORD_SELECT 0x0000001A
+#define AC97_1C_RECORD_GAIN 0x0000001C
+#define AC97_1E_RESERVED_1E 0x0000001E
+#define AC97_20_GENERAL_PURPOSE 0x00000020
+#define AC97_22_3D_CONTROL 0x00000022
+#define AC97_24_MODEM_RATE 0x00000024
+#define AC97_26_POWERDOWN 0x00000026
+#define AC97_28_EXT_AUDIO_ID 0x00000028
+#define AC97_2A_EXT_AUDIO_POWER 0x0000002A
+#define AC97_2C_PCM_FRONT_DAC_RATE 0x0000002C
+#define AC97_2E_PCM_SURR_DAC_RATE 0x0000002E
+#define AC97_30_PCM_LFE_DAC_RATE 0x00000030
+#define AC97_32_PCM_LR_ADC_RATE 0x00000032
+#define AC97_34_MIC_ADC_RATE 0x00000034
+#define AC97_36_6CH_VOL_C_LFE 0x00000036
+#define AC97_38_6CH_VOL_SURROUND 0x00000038
+#define AC97_3A_SPDIF_CONTROL 0x0000003A
+#define AC97_3C_EXT_MODEM_ID 0x0000003C
+#define AC97_3E_EXT_MODEM_POWER 0x0000003E
+#define AC97_40_LINE1_CODEC_RATE 0x00000040
+#define AC97_42_LINE2_CODEC_RATE 0x00000042
+#define AC97_44_HANDSET_CODEC_RATE 0x00000044
+#define AC97_46_LINE1_CODEC_LEVEL 0x00000046
+#define AC97_48_LINE2_CODEC_LEVEL 0x00000048
+#define AC97_4A_HANDSET_CODEC_LEVEL 0x0000004A
+#define AC97_4C_GPIO_PIN_CONFIG 0x0000004C
+#define AC97_4E_GPIO_PIN_TYPE 0x0000004E
+#define AC97_50_GPIO_PIN_STICKY 0x00000050
+#define AC97_52_GPIO_PIN_WAKEUP 0x00000052
+#define AC97_54_GPIO_PIN_STATUS 0x00000054
+#define AC97_56_RESERVED 0x00000056
+#define AC97_58_RESERVED 0x00000058
+#define AC97_5A_CRYSTAL_REV_N_FAB_ID 0x0000005A
+#define AC97_5C_TEST_AND_MISC_CTRL 0x0000005C
+#define AC97_5E_AC_MODE 0x0000005E
+#define AC97_60_MISC_CRYSTAL_CONTROL 0x00000060
+#define AC97_62_VENDOR_RESERVED 0x00000062
+#define AC97_64_DAC_SRC_PHASE_INCR 0x00000064
+#define AC97_66_ADC_SRC_PHASE_INCR 0x00000066
+#define AC97_68_RESERVED_68 0x00000068
+#define AC97_6A_SERIAL_PORT_CONTROL 0x0000006A
+#define AC97_6C_VENDOR_RESERVED 0x0000006C
+#define AC97_6E_VENDOR_RESERVED 0x0000006E
+#define AC97_70_BDI_CONFIG 0x00000070
+#define AC97_72_BDI_WAKEUP 0x00000072
+#define AC97_74_VENDOR_RESERVED 0x00000074
+#define AC97_76_CAL_ADDRESS 0x00000076
+#define AC97_78_CAL_DATA 0x00000078
+#define AC97_7A_VENDOR_RESERVED 0x0000007A
+#define AC97_7C_VENDOR_ID1 0x0000007C
+#define AC97_7E_VENDOR_ID2 0x0000007E
+
+
+#ifndef __ASSEMBLY__
+
+//
+// enum type for use with reg AC97_RECORD_SELECT
+//
+typedef enum{
+ RECORD_MIC = 0x0000,
+ RECORD_CD = 0x0101,
+ RECORD_VIDEO_IN = 0x0202,
+ RECORD_AUX_IN = 0x0303,
+ RECORD_LINE_IN = 0x0404,
+ RECORD_STEREO_MIX = 0x0505,
+ RECORD_MONO_MIX = 0x0606,
+ RECORD_PHONE_IN = 0x0707
+} Ac97RecordSources;
+
+#endif /* __ASSEMBLY__ */
+
+//
+// Sample rates supported directly in AC97_PCM_FRONT_DAC_RATE and
+// AC97_PCM_LR_ADC_RATE.
+//
+#define Ac97_Fs_8000 0x1f40
+#define Ac97_Fs_11025 0x2b11
+#define Ac97_Fs_16000 0x3e80
+#define Ac97_Fs_22050 0x5622
+#define Ac97_Fs_32000 0x7d00
+#define Ac97_Fs_44100 0xac44
+#define Ac97_Fs_48000 0xbb80
+
+//
+// RSIZE and TSIZE in AC97RXCR and AC97TXCR
+//
+#define Ac97_SIZE_20 2
+#define Ac97_SIZE_18 1
+#define Ac97_SIZE_16 0
+#define Ac97_SIZE_12 3
+
+//=============================================================================
+//=============================================================================
+
+
+#endif /* _REGS_AC97_H_ */
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -11,6 +11,23 @@ menuconfig SND_ARM
if SND_ARM
+config SND_EP93XX_AC97
+ tristate "AC97 driver for the Cirrus EP93xx chip"
+ depends on ARCH_EP93XX && SND
+ select SND_EP93XX_PCM
+ select SND_AC97_CODEC
+ help
+ Say Y here to use AC'97 audio with a Cirrus Logic EP93xx chip.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ep93xx-ac97.
+
+config SND_EP93XX_PCM
+ tristate
+ select SND_PCM
+ help
+ Generic PCM module for EP93xx
+
config SND_ARMAACI
tristate "ARM PrimeCell PL041 AC Link support"
depends on ARM_AMBA
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -5,6 +5,9 @@
obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
snd-aaci-objs := aaci.o devdma.o
+obj-$(CONFIG_SND_EP93XX_AC97) += snd-ep93xx-ac97.o
+snd-ep93xx-ac97-objs := ep93xx-ac97.o
+
obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o
snd-pxa2xx-pcm-objs := pxa2xx-pcm.o
--- /dev/null
+++ b/sound/arm/ep93xx-ac97.c
@@ -0,0 +1,3482 @@
+/*
+ * linux/sound/arm/ep93xx-ac97.c -- ALSA PCM interface for the edb93xx ac97 audio
+ */
+
+#include <linux/autoconf.h>
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/soundcard.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include <asm/irq.h>
+#include <asm/semaphore.h>
+#include <asm/hardware.h>
+#include <asm/io.h>
+#include <asm/arch/dma.h>
+#include "ep93xx-ac97.h"
+
+#define DRIVER_VERSION "01/05/2009"
+#define DRIVER_DESC "EP93xx AC97 Audio driver"
+static int ac_link_enabled = 0;
+static int codec_supported_mixers;
+
+//#define DEBUG 1
+#ifdef DEBUG
+#define DPRINTK( fmt, arg... ) printk( fmt, ##arg )
+#else
+#define DPRINTK( fmt, arg... )
+#endif
+
+#define WL16 0
+#define WL24 1
+
+#define AUDIO_NAME "ep93xx-ac97"
+#define AUDIO_SAMPLE_RATE_DEFAULT 44100
+#define AUDIO_DEFAULT_VOLUME 0
+#define AUDIO_MAX_VOLUME 181
+#define AUDIO_DEFAULT_DMACHANNELS 3
+#define PLAYBACK_DEFAULT_DMACHANNELS 3
+#define CAPTURE_DEFAULT_DMACHANNELS 3
+
+#define CHANNEL_FRONT (1<<0)
+#define CHANNEL_REAR (1<<1)
+#define CHANNEL_CENTER_LFE (1<<2)
+
+static void snd_ep93xx_dma_tx_callback( ep93xx_dma_int_t DMAInt,
+ ep93xx_dma_dev_t device,
+ unsigned int user_data);
+static void snd_ep93xx_dma_rx_callback( ep93xx_dma_int_t DMAInt,
+ ep93xx_dma_dev_t device,
+ unsigned int user_data);
+
+static const struct snd_pcm_hardware ep93xx_ac97_pcm_hardware = {
+
+
+ .info = ( SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE ),
+ .formats = ( SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_BE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_U32_BE | SNDRV_PCM_FMTBIT_S32_BE ),
+ .rates = ( SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 ),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,/*2,*/
+ .channels_max = 2,
+
+ .period_bytes_min = 1 * 1024,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 1,
+ .periods_max = 32,
+ .buffer_bytes_max = 32 * 1024,
+ .fifo_size = 0,
+};
+
+static audio_stream_t output_stream;
+static audio_stream_t input_stream;
+
+static audio_state_t audio_state =
+{
+ .output_stream =&output_stream,
+ .output_dma[0] =DMATx_AAC1,
+ .output_id[0] ="Ac97 out",
+
+ .input_stream =&input_stream,
+ .input_dma[0] =DMARx_AAC1,
+ .input_id[0] ="Ac97 in",
+
+ .sem = __SEMAPHORE_INIT(audio_state.sem,1),
+ .codec_set_by_playback = 0,
+ .codec_set_by_capture = 0,
+ .DAC_bit_width =16,
+ .bCompactMode =0,
+};
+
+
+
+/*
+ * peek
+ *
+ * Reads an AC97 codec register. Returns -1 if there was an error.
+ */
+static int peek(unsigned int uiAddress)
+{
+ unsigned int uiAC97RGIS;
+
+ if( !ac_link_enabled )
+ {
+ printk("ep93xx ac97 peek: attempt to peek before enabling ac-link.\n");
+ return -1;
+ }
+
+ /*
+ * Check to make sure that the address is aligned on a word boundary
+ * and is 7E or less.
+ */
+ if( ((uiAddress & 0x1)!=0) || (uiAddress > 0x007e))
+ {
+ return -1;
+ }
+
+ /*
+ * How it is supposed to work is:
+ * - The ac97 controller sends out a read addr in slot 1.
+ * - In the next frame, the codec will echo that address back in slot 1
+ * and send the data in slot 2. SLOT2RXVALID will be set to 1.
+ *
+ * Read until SLOT2RXVALID goes to 1. Reading the data in AC97S2DATA
+ * clears SLOT2RXVALID.
+ */
+
+ /*
+ * First, delay one frame in case of back to back peeks/pokes.
+ */
+ mdelay( 1 );
+
+ /*
+ * Write the address to AC97S1DATA, delay 1 frame, read the flags.
+ */
+ outl( uiAddress, AC97S1DATA);
+ udelay( 21 * 4 );
+ uiAC97RGIS = inl( AC97RGIS );
+
+ /*
+ * Return error if we timed out.
+ */
+ if( ((uiAC97RGIS & AC97RGIS_SLOT1TXCOMPLETE) == 0 ) &&
+ ((uiAC97RGIS & AC97RGIS_SLOT2RXVALID) == 0 ) )
+ {
+ printk( "ep93xx-ac97 - peek failed reading reg 0x%02x.\n", uiAddress );
+ return -1;
+ }
+
+ return ( inl(AC97S2DATA) & 0x000fffff);
+}
+
+/*
+ * poke
+ *
+ * Writes an AC97 codec Register. Return -1 if error.
+ */
+static int poke(unsigned int uiAddress, unsigned int uiValue)
+{
+ unsigned int uiAC97RGIS;
+
+ if( !ac_link_enabled )
+ {
+ printk("ep93xx ac97 poke: attempt to poke before enabling ac-link.\n");
+ return -1;
+ }
+
+ /*
+ * Check to make sure that the address is align on a word boundary and
+ * is 7E or less. And that the value is a 16 bit value.
+ */
+ if( ((uiAddress & 0x1)!=0) || (uiAddress > 0x007e))
+ {
+ printk("ep93xx ac97 poke: address error.\n");
+ return -1;
+ }
+
+ /*stop the audio loop from the input to the output directly*/
+
+ if((uiAddress==AC97_0E_MIC_VOL)||(uiAddress==AC97_10_LINE_IN_VOL))
+ {
+ uiValue = (uiValue | 0x8000);
+
+ }
+
+ /*
+ * First, delay one frame in case of back to back peeks/pokes.
+ */
+ mdelay( 1 );
+
+ /*
+ * Write the data to AC97S2DATA, then the address to AC97S1DATA.
+ */
+ outl( uiValue, AC97S2DATA );
+ outl( uiAddress, AC97S1DATA );
+
+ /*
+ * Wait for the tx to complete, get status.
+ */
+ udelay( 30 );/*21*/
+ uiAC97RGIS = inl(AC97RGIS);
+
+ /*
+ * Return error if we timed out.
+ */
+ if( !(inl(AC97RGIS) & AC97RGIS_SLOT1TXCOMPLETE) )
+ {
+ printk( "ep93xx-ac97: poke failed writing reg 0x%02x value 0x%02x.\n", uiAddress, uiValue );
+ return -1;
+ }
+
+ return 0;
+}
+
+
+/*
+ * When we get to the multichannel case the pre-fill and enable code
+ * will go to the dma driver's start routine.
+ */
+static void ep93xx_audio_enable( int input_or_output_stream )
+{
+ unsigned int uiTemp;
+
+ DPRINTK("ep93xx_audio_enable :%x\n",input_or_output_stream);
+
+ /*
+ * Enable the rx or tx channel depending on the value of
+ * input_or_output_stream
+ */
+ if( input_or_output_stream )
+ {
+ uiTemp = inl(AC97TXCR1);
+ outl( (uiTemp | AC97TXCR_TEN), AC97TXCR1 );
+ }
+ else
+ {
+ uiTemp = inl(AC97RXCR1);
+ outl( (uiTemp | AC97RXCR_REN), AC97RXCR1 );
+ }
+
+
+ //DDEBUG("ep93xx_audio_enable - EXIT\n");
+}
+
+static void ep93xx_audio_disable( int input_or_output_stream )
+{
+ unsigned int uiTemp;
+
+ DPRINTK("ep93xx_audio_disable\n");
+
+ /*
+ * Disable the rx or tx channel depending on the value of
+ * input_or_output_stream
+ */
+ if( input_or_output_stream )
+ {
+ uiTemp = inl(AC97TXCR1);
+ outl( (uiTemp & ~AC97TXCR_TEN), AC97TXCR1 );
+ }
+ else
+ {
+ uiTemp = inl(AC97RXCR1);
+ outl( (uiTemp & ~AC97RXCR_REN), AC97RXCR1 );
+ }
+
+ //DDEBUG("ep93xx_audio_disable - EXIT\n");
+}
+
+
+
+/*=======================================================================================*/
+/*
+ * ep93xx_setup_src
+ *
+ * Once the ac-link is up and all is good, we want to set the codec to a
+ * usable mode.
+ */
+static void ep93xx_setup_src(void)
+{
+ int iTemp;
+
+ /*
+ * Set the VRA bit to enable the SRC.
+ */
+ iTemp = peek( AC97_2A_EXT_AUDIO_POWER );
+ poke( AC97_2A_EXT_AUDIO_POWER, (iTemp | 0x1) );
+
+ /*
+ * Set the DSRC/ASRC bits to enable the variable rate SRC.
+ */
+ iTemp = peek( AC97_60_MISC_CRYSTAL_CONTROL );
+ poke( AC97_60_MISC_CRYSTAL_CONTROL, (iTemp | 0x0300) );
+}
+
+/*
+ * ep93xx_set_samplerate
+ *
+ * lFrequency - Sample Rate in Hz
+ * bCapture - 0 to set Tx sample rate; 1 to set Rx sample rate
+ */
+static void ep93xx_set_samplerate( long lSampleRate, int bCapture )
+{
+ unsigned short usDivider, usPhase;
+
+ DPRINTK( "ep93xx_set_samplerate - Fs = %d\n", (int)lSampleRate );
+
+ if( (lSampleRate < 7200) || (lSampleRate > 48000) )
+ {
+ printk( "ep93xx_set_samplerate - invalid Fs = %d\n",
+ (int)lSampleRate );
+ return;
+ }
+
+ /*
+ * Calculate divider and phase increment.
+ *
+ * divider = round( 0x1770000 / lSampleRate )
+ * Note that usually rounding is done by adding 0.5 to a floating
+ * value and then truncating. To do this without using floating
+ * point, I multiply the fraction by two, do the division, then add one,
+ * then divide the whole by 2 and then truncate.
+ * Same effect, no floating point math.
+ *
+ * Ph incr = trunc( (0x1000000 / usDivider) + 1 )
+ */
+
+ usDivider = (unsigned short)( ((2 * 0x1770000 / lSampleRate) + 1) / 2 );
+
+ usPhase = (0x1000000 / usDivider) + 1;
+
+ /*
+ * Write them in the registers. Spec says divider must be
+ * written after phase incr.
+ */
+ if(!bCapture)
+ {
+ poke( AC97_2C_PCM_FRONT_DAC_RATE, usDivider);
+ poke( AC97_64_DAC_SRC_PHASE_INCR, usPhase);
+ }
+ else
+ {
+
+ poke( AC97_32_PCM_LR_ADC_RATE, usDivider);
+ poke( AC97_66_ADC_SRC_PHASE_INCR, usPhase);
+ }
+
+ DPRINTK( "ep93xx_set_samplerate - phase = %d, divider = %d\n",
+ (unsigned int)usPhase, (unsigned int)usDivider );
+
+ /*
+ * We sorta should report the actual samplerate back to the calling
+ * application. But some applications freak out if they don't get
+ * exactly what they asked for. So we fudge and tell them what
+ * they want to hear.
+ */
+ //audio_samplerate = lSampleRate;
+
+ DPRINTK( "ep93xx_set_samplerate - EXIT\n" );
+}
+
+/*
+ * ep93xx_set_hw_format
+ *
+ * Sets up whether the controller is expecting 20 bit data in 32 bit words
+ * or 16 bit data compacted to have a stereo sample in each 32 bit word.
+ */
+static void ep93xx_set_hw_format( long format,long channel )
+{
+ int bCompactMode;
+
+ switch( format )
+ {
+ /*
+ * Here's all the <=16 bit formats. We can squeeze both L and R
+ * into one 32 bit sample so use compact mode.
+ */
+ case SNDRV_PCM_FORMAT_U8:
+ case SNDRV_PCM_FORMAT_S8:
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_U16_LE:
+ bCompactMode = 1;
+ break;
+
+ /*
+ * Add any other >16 bit formats here...
+ */
+ case SNDRV_PCM_FORMAT_S32_LE:
+ default:
+ bCompactMode = 0;
+ break;
+ }
+
+ if( bCompactMode )
+ {
+ DPRINTK("ep93xx_set_hw_format - Setting serial mode to 16 bit compact.\n");
+
+ /*
+ * Turn on Compact Mode so we can fit each stereo sample into
+ * a 32 bit word. Twice as efficent for DMA and FIFOs.
+ */
+ if(channel==2){
+ outl( 0x00008018, AC97RXCR1 );
+ outl( 0x00008018, AC97TXCR1 );
+ }
+ else {
+ outl( 0x00008018, AC97RXCR1 );
+ outl( 0x00008018, AC97TXCR1 );
+ }
+
+
+ audio_state.DAC_bit_width = 16;
+ audio_state.bCompactMode = 1;
+ }
+ else
+ {
+ DPRINTK("ep93xx_set_hw_format - Setting serial mode to 20 bit non-CM.\n");
+
+ /*
+ * Turn off Compact Mode so we can do > 16 bits per channel
+ */
+ if(channel==2){
+ outl( 0x00004018, AC97RXCR1 );
+ outl( 0x00004018, AC97TXCR1 );
+ }
+ else{
+ outl( 0x00004018, AC97RXCR1 );
+ outl( 0x00004018, AC97TXCR1 );
+ }
+
+ audio_state.DAC_bit_width = 20;
+ audio_state.bCompactMode = 0;
+ }
+
+}
+
+/*
+ * ep93xx_stop_loop
+ *
+ * Once the ac-link is up and all is good, we want to set the codec to a
+ * usable mode.
+ */
+static void ep93xx_stop_loop(void)
+{
+ int iTemp;
+
+ /*
+ * Set the AC97_0E_MIC_VOL MUTE bit to enable the LOOP.
+ */
+ iTemp = peek( AC97_0E_MIC_VOL );
+ poke( AC97_0E_MIC_VOL, (iTemp | 0x8000) );
+
+ /*
+ * Set the AC97_10_LINE_IN_VOL MUTE bit to enable the LOOP.
+ */
+ iTemp = peek( AC97_10_LINE_IN_VOL );
+ poke( AC97_10_LINE_IN_VOL, (iTemp | 0x8000) );
+}
+
+/*
+ * ep93xx_init_ac97_controller
+ *
+ * This routine sets up the Ac'97 Controller.
+ */
+static void ep93xx_init_ac97_controller(void)
+{
+ unsigned int uiDEVCFG, uiTemp;
+
+ DPRINTK("ep93xx_init_ac97_controller - enter\n");
+
+ /*
+ * Configure the multiplexed Ac'97 pins to be Ac97 not I2s.
+ * Configure the EGPIO4 and EGPIO6 to be GPIOS, not to be
+ * SDOUT's for the second and third I2S controller channels.
+ */
+ uiDEVCFG = inl(EP93XX_SYSCON_DEVICE_CONFIG);
+
+ uiDEVCFG &= ~(EP93XX_SYSCON_DEVCFG_CONFIG_I2SONAC97 |
+ EP93XX_SYSCON_DEVCFG_A1onG |
+ EP93XX_SYSCON_DEVCFG_A2onG);
+
+ SysconSetLocked(EP93XX_SYSCON_DEVICE_CONFIG, uiDEVCFG);
+
+ /*
+ * Disable the AC97 controller internal loopback.
+ * Disable Override codec ready.
+ */
+ outl( 0, AC97GCR );
+
+ /*
+ * Enable the AC97 Link.
+ */
+ uiTemp = inl(AC97GCR);
+ outl( (uiTemp | AC97GSR_IFE), AC97GCR );
+
+ /*
+ * Set the TIMEDRESET bit. Will cause a > 1uSec reset of the ac-link.
+ * This bit is self resetting.
+ */
+ outl( AC97RESET_TIMEDRESET, AC97RESET );
+
+ /*
+ * Delay briefly, but let's not hog the processor.
+ */
+ set_current_state(TASK_INTERRUPTIBLE);
+ schedule_timeout( 5 ); /* 50 mSec */
+
+ /*
+ * Read the AC97 status register to see if we've seen a CODECREADY
+ * signal from the AC97 codec.
+ */
+ if( !(inl(AC97RGIS) & AC97RGIS_CODECREADY))
+ {
+ printk( "ep93xx-ac97 - FAIL: CODECREADY still low!\n");
+ return;
+ }
+
+ /*
+ * Delay for a second, not hogging the processor
+ */
+ set_current_state(TASK_INTERRUPTIBLE);
+ schedule_timeout( HZ ); /* 1 Sec */
+
+ /*
+ * Now the Ac-link is up. We can read and write codec registers.
+ */
+ ac_link_enabled = 1;
+
+ /*
+ * Set up the rx and tx channels
+ * Set the CM bit, data size=16 bits, enable tx slots 3 & 4.
+ */
+ ep93xx_set_hw_format( EP93XX_DEFAULT_FORMAT,EP93XX_DEFAULT_NUM_CHANNELS );
+
+ DPRINTK( "ep93xx-ac97 -- AC97RXCR1: %08x\n", inl(AC97RXCR1) );
+ DPRINTK( "ep93xx-ac97 -- AC97TXCR1: %08x\n", inl(AC97TXCR1) );
+
+ DPRINTK("ep93xx_init_ac97_controller - EXIT - success\n");
+
+}
+
+#ifdef alsa_ac97_debug
+static void ep93xx_dump_ac97_regs(void)
+{
+ int i;
+ unsigned int reg0, reg1, reg2, reg3, reg4, reg5, reg6, reg7;
+
+ DPRINTK( "---------------------------------------------\n");
+ DPRINTK( " : 0 2 4 6 8 A C E\n" );
+
+ for( i=0 ; i < 0x80 ; i+=0x10 )
+ {
+ reg0 = 0xffff & (unsigned int)peek( i );
+ reg1 = 0xffff & (unsigned int)peek( i + 0x2 );
+ reg2 = 0xffff & (unsigned int)peek( i + 0x4 );
+ reg3 = 0xffff & (unsigned int)peek( i + 0x6 );
+ reg4 = 0xffff & (unsigned int)peek( i + 0x8 );
+ reg5 = 0xffff & (unsigned int)peek( i + 0xa );
+ reg6 = 0xffff & (unsigned int)peek( i + 0xc );
+ reg7 = 0xffff & (unsigned int)peek( i + 0xe );
+
+ DPRINTK( " %02x : %04x %04x %04x %04x %04x %04x %04x %04x\n",
+ i, reg0, reg1, reg2, reg3, reg4, reg5, reg6, reg7);
+ }
+
+ DPRINTK( "---------------------------------------------\n");
+}
+#endif
+
+
+#define supported_mixer(FOO) \
+ ( (FOO >= 0) && \
+ (FOO < SOUND_MIXER_NRDEVICES) && \
+ codec_supported_mixers & (1<<FOO) )
+
+/*
+ * Available record sources.
+ * LINE1 refers to AUX in.
+ * IGAIN refers to input gain which means stereo mix.
+ */
+#define AC97_RECORD_MASK \
+ (SOUND_MASK_MIC | SOUND_MASK_CD | SOUND_MASK_IGAIN | SOUND_MASK_VIDEO |\
+ SOUND_MASK_LINE1 | SOUND_MASK_LINE | SOUND_MASK_PHONEIN)
+
+#define AC97_STEREO_MASK \
+ (SOUND_MASK_VOLUME | SOUND_MASK_PCM | SOUND_MASK_LINE | SOUND_MASK_CD | \
+ SOUND_MASK_ALTPCM | SOUND_MASK_IGAIN | SOUND_MASK_LINE1 | SOUND_MASK_VIDEO)
+
+#define AC97_SUPPORTED_MASK \
+ (AC97_STEREO_MASK | SOUND_MASK_BASS | SOUND_MASK_TREBLE | \
+ SOUND_MASK_SPEAKER | SOUND_MASK_MIC | \
+ SOUND_MASK_PHONEIN | SOUND_MASK_PHONEOUT)
+
+
+
+
+/* this table has default mixer values for all OSS mixers. */
+typedef struct {
+ int mixer;
+ unsigned int value;
+} mixer_defaults_t;
+
+/*
+ * Default mixer settings that are set up during boot.
+ *
+ * These values are 16 bit numbers in which the upper byte is right volume
+ * and the lower byte is left volume or mono volume for mono controls.
+ *
+ * OSS Range for each of left and right volumes is 0 to 100 (0x00 to 0x64).
+ *
+ */
+static mixer_defaults_t mixer_defaults[SOUND_MIXER_NRDEVICES] =
+{
+ /* Outputs */
+ {SOUND_MIXER_VOLUME, 0x6464}, /* 0 dB */ /* -46.5dB to 0 dB */
+ {SOUND_MIXER_ALTPCM, 0x6464}, /* 0 dB */ /* -46.5dB to 0 dB */
+ {SOUND_MIXER_PHONEOUT, 0x6464}, /* 0 dB */ /* -46.5dB to 0 dB */
+
+ /* PCM playback gain */
+ {SOUND_MIXER_PCM, 0x4b4b}, /* 0 dB */ /* -34.5dB to +12dB */
+
+ /* Record gain */
+ {SOUND_MIXER_IGAIN, 0x0000}, /* 0 dB */ /* -34.5dB to +12dB */
+
+ /* Inputs */
+ {SOUND_MIXER_MIC, 0x0000}, /* mute */ /* -34.5dB to +12dB */
+ {SOUND_MIXER_LINE, 0x4b4b}, /* 0 dB */ /* -34.5dB to +12dB */
+
+ /* Inputs that are not connected. */
+ {SOUND_MIXER_SPEAKER, 0x0000}, /* mute */ /* -45dB to 0dB */
+ {SOUND_MIXER_PHONEIN, 0x0000}, /* mute */ /* -34.5dB to +12dB */
+ {SOUND_MIXER_CD, 0x0000}, /* mute */ /* -34.5dB to +12dB */
+ {SOUND_MIXER_VIDEO, 0x0000}, /* mute */ /* -34.5dB to +12dB */
+ {SOUND_MIXER_LINE1, 0x0000}, /* mute */ /* -34.5dB to +12dB */
+
+ {-1,0} /* last entry */
+};
+
+/* table to scale scale from OSS mixer value to AC97 mixer register value */
+typedef struct {
+ unsigned int offset;
+ int scale;
+} ac97_mixer_hw_t;
+
+static ac97_mixer_hw_t ac97_hw[SOUND_MIXER_NRDEVICES] =
+{
+ [SOUND_MIXER_VOLUME] = {AC97_02_MASTER_VOL, 64},
+ [SOUND_MIXER_BASS] = {0, 0},
+ [SOUND_MIXER_TREBLE] = {0, 0},
+ [SOUND_MIXER_SYNTH] = {0, 0},
+ [SOUND_MIXER_PCM] = {AC97_18_PCM_OUT_VOL, 32},
+ [SOUND_MIXER_SPEAKER] = {AC97_0A_PC_BEEP_VOL, 32},
+ [SOUND_MIXER_LINE] = {AC97_10_LINE_IN_VOL, 32},
+ [SOUND_MIXER_MIC] = {AC97_0E_MIC_VOL, 32},
+ [SOUND_MIXER_CD] = {AC97_12_CD_VOL, 32},
+ [SOUND_MIXER_IMIX] = {0, 0},
+ [SOUND_MIXER_ALTPCM] = {AC97_04_HEADPHONE_VOL, 64},
+ [SOUND_MIXER_RECLEV] = {0, 0},
+ [SOUND_MIXER_IGAIN] = {AC97_1C_RECORD_GAIN, 16},
+ [SOUND_MIXER_OGAIN] = {0, 0},
+ [SOUND_MIXER_LINE1] = {AC97_16_AUX_VOL, 32},
+ [SOUND_MIXER_LINE2] = {0, 0},
+ [SOUND_MIXER_LINE3] = {0, 0},
+ [SOUND_MIXER_DIGITAL1] = {0, 0},
+ [SOUND_MIXER_DIGITAL2] = {0, 0},
+ [SOUND_MIXER_DIGITAL3] = {0, 0},
+ [SOUND_MIXER_PHONEIN] = {AC97_0C_PHONE_VOL, 32},
+ [SOUND_MIXER_PHONEOUT] = {AC97_06_MONO_VOL, 64},
+ [SOUND_MIXER_VIDEO] = {AC97_14_VIDEO_VOL, 32},
+ [SOUND_MIXER_RADIO] = {0, 0},
+ [SOUND_MIXER_MONITOR] = {0, 0},
+};
+
+
+/* the following tables allow us to go from OSS <-> ac97 quickly. */
+enum ac97_recsettings
+{
+ AC97_REC_MIC=0,
+ AC97_REC_CD,
+ AC97_REC_VIDEO,
+ AC97_REC_AUX,
+ AC97_REC_LINE,
+ AC97_REC_STEREO, /* combination of all enabled outputs.. */
+ AC97_REC_MONO, /*.. or the mono equivalent */
+ AC97_REC_PHONE
+};
+
+static const unsigned int ac97_rm2oss[] =
+{
+ [AC97_REC_MIC] = SOUND_MIXER_MIC,
+ [AC97_REC_CD] = SOUND_MIXER_CD,
+ [AC97_REC_VIDEO] = SOUND_MIXER_VIDEO,
+ [AC97_REC_AUX] = SOUND_MIXER_LINE1,
+ [AC97_REC_LINE] = SOUND_MIXER_LINE,
+ [AC97_REC_STEREO]= SOUND_MIXER_IGAIN,
+ [AC97_REC_PHONE] = SOUND_MIXER_PHONEIN
+};
+
+/* indexed by bit position */
+static const unsigned int ac97_oss_rm[] =
+{
+ [SOUND_MIXER_MIC] = AC97_REC_MIC,
+ [SOUND_MIXER_CD] = AC97_REC_CD,
+ [SOUND_MIXER_VIDEO] = AC97_REC_VIDEO,
+ [SOUND_MIXER_LINE1] = AC97_REC_AUX,
+ [SOUND_MIXER_LINE] = AC97_REC_LINE,
+ [SOUND_MIXER_IGAIN] = AC97_REC_STEREO,
+ [SOUND_MIXER_PHONEIN] = AC97_REC_PHONE
+};
+
+
+/*
+ * ep93xx_write_mixer
+ *
+ */
+static void ep93xx_write_mixer
+(
+ int oss_channel,
+ unsigned int left,
+ unsigned int right
+)
+{
+ u16 val = 0;
+ ac97_mixer_hw_t * mh = &ac97_hw[oss_channel];
+
+ DPRINTK("ac97_codec: wrote OSS %2d (ac97 0x%02x), "
+ "l:%2d, r:%2d:",
+ oss_channel, mh->offset, left, right);
+
+ if( !mh->scale )
+ {
+ printk( "ep93xx-ac97.c: ep93xx_write_mixer - not a valid OSS channel\n");
+ return;
+ }
+
+ if( AC97_STEREO_MASK & (1 << oss_channel) )
+ {
+ /* stereo mixers */
+ if (left == 0 && right == 0)
+ {
+ val = 0x8000;
+ }
+ else
+ {
+ if (oss_channel == SOUND_MIXER_IGAIN)
+ {
+ right = (right * mh->scale) / 100;
+ left = (left * mh->scale) / 100;
+ if (right >= mh->scale)
+ right = mh->scale-1;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ }
+ else
+ {
+ right = ((100 - right) * mh->scale) / 100;
+ left = ((100 - left) * mh->scale) / 100;
+ if (right >= mh->scale)
+ right = mh->scale-1;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ }
+ val = (left << 8) | right;
+ }
+ }
+ else if(left == 0)
+ {
+ val = 0x8000;
+ }
+ else if( (oss_channel == SOUND_MIXER_SPEAKER) ||
+ (oss_channel == SOUND_MIXER_PHONEIN) ||
+ (oss_channel == SOUND_MIXER_PHONEOUT) )
+ {
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val = left;
+ }
+ else if (oss_channel == SOUND_MIXER_MIC)
+ {
+ val = peek( mh->offset) & ~0x801f;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= left;
+ }
+ /*
+ * For bass and treble, the low bit is optional. Masking it
+ * lets us avoid the 0xf 'bypass'.
+ * Do a read, modify, write as we have two contols in one reg.
+ */
+ else if (oss_channel == SOUND_MIXER_BASS)
+ {
+ val = peek( mh->offset) & ~0x0f00;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= (left << 8) & 0x0e00;
+ }
+ else if (oss_channel == SOUND_MIXER_TREBLE)
+ {
+ val = peek( mh->offset) & ~0x000f;
+ left = ((100 - left) * mh->scale) / 100;
+ if (left >= mh->scale)
+ left = mh->scale-1;
+ val |= left & 0x000e;
+ }
+
+ DPRINTK(" 0x%04x", val);
+
+ poke( mh->offset, val );
+
+#ifdef alsa_ac97_debug
+ val = peek( mh->offset );
+ DEBUG(" -> 0x%04x\n", val);
+#endif
+
+}
+
+/* a thin wrapper for write_mixer */
+static void ep93xx_set_mixer
+(
+ unsigned int oss_mixer,
+ unsigned int val
+)
+{
+ unsigned int left,right;
+
+ /* cleanse input a little */
+ right = ((val >> 8) & 0xff) ;
+ left = (val & 0xff) ;
+
+ if (right > 100) right = 100;
+ if (left > 100) left = 100;
+
+ /*mixer_state[oss_mixer] = (right << 8) | left;*/
+ ep93xx_write_mixer( oss_mixer, left, right);
+}
+
+static void ep93xx_init_mixer(void)
+{
+ u16 cap;
+ int i;
+
+ /* mixer masks */
+ codec_supported_mixers = AC97_SUPPORTED_MASK;
+
+ cap = peek( AC97_00_RESET );
+ if( !(cap & 0x04) )
+ {
+ codec_supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE);
+ }
+ if( !(cap & 0x10) )
+ {
+ codec_supported_mixers &= ~SOUND_MASK_ALTPCM;
+ }
+
+ /*
+ * Detect bit resolution of output volume controls by writing to the
+ * 6th bit (not unmuting yet)
+ */
+ poke( AC97_02_MASTER_VOL, 0xa020 );
+ if( peek( AC97_02_MASTER_VOL) != 0xa020 )
+ {
+ ac97_hw[SOUND_MIXER_VOLUME].scale = 32;
+ }
+
+ poke( AC97_04_HEADPHONE_VOL, 0xa020 );
+ if( peek( AC97_04_HEADPHONE_VOL) != 0xa020 )
+ {
+ ac97_hw[AC97_04_HEADPHONE_VOL].scale = 32;
+ }
+
+ poke( AC97_06_MONO_VOL, 0x8020 );
+ if( peek( AC97_06_MONO_VOL) != 0x8020 )
+ {
+ ac97_hw[AC97_06_MONO_VOL].scale = 32;
+ }
+
+ /* initialize mixer channel volumes */
+ for( i = 0;
+ (i < SOUND_MIXER_NRDEVICES) && (mixer_defaults[i].mixer != -1) ;
+ i++ )
+ {
+ if( !supported_mixer( mixer_defaults[i].mixer) )
+ {
+ continue;
+ }
+
+ ep93xx_set_mixer( mixer_defaults[i].mixer, mixer_defaults[i].value);
+ }
+
+}
+
+static int ep93xx_set_recsource( int mask )
+{
+ unsigned int val;
+
+ /* Arg contains a bit for each recording source */
+ if( mask == 0 )
+ {
+ return 0;
+ }
+
+ mask &= AC97_RECORD_MASK;
+
+ if( mask == 0 )
+ {
+ return -EINVAL;
+ }
+
+ /*
+ * May have more than one bit set. So clear out currently selected
+ * record source value first (AC97 supports only 1 input)
+ */
+ val = (1 << ac97_rm2oss[peek( AC97_1A_RECORD_SELECT ) & 0x07]);
+ if (mask != val)
+ mask &= ~val;
+
+ val = ffs(mask);
+ val = ac97_oss_rm[val-1];
+ val |= val << 8; /* set both channels */
+
+ /*
+ *
+ */
+ val = peek( AC97_1A_RECORD_SELECT ) & 0x0707;
+ if ((val&0x0404)!=0)
+ val=0x0404;
+ else if((val&0x0000)!=0)
+ val=0x0101;
+
+
+ DPRINTK("ac97_codec: setting ac97 recmask to 0x%04x\n", val);
+
+ poke( AC97_1A_RECORD_SELECT, val);
+
+ return 0;
+}
+
+/*
+ * ep93xx_init_ac97_codec
+ *
+ * Program up the external Ac97 codec.
+ *
+ */
+static void ep93xx_init_ac97_codec( void )
+{
+ DPRINTK("ep93xx_init_ac97_codec - enter\n");
+
+ ep93xx_setup_src();
+ ep93xx_set_samplerate( AUDIO_SAMPLE_RATE_DEFAULT, 0 );
+ ep93xx_set_samplerate( AUDIO_SAMPLE_RATE_DEFAULT, 1 );
+ ep93xx_init_mixer();
+
+ DPRINTK("ep93xx_init_ac97_codec - EXIT\n");
+
+}
+
+
+/*
+ * ep93xx_audio_init
+ * Audio interface
+ */
+static void ep93xx_audio_init(void)
+{
+ DPRINTK("ep93xx_audio_init - enter\n");
+ /*
+ * Init the controller, enable the ac-link.
+ * Initialize the codec.
+ */
+ ep93xx_init_ac97_controller();
+ ep93xx_init_ac97_codec();
+ /*stop the audio loop from the input to the output directly*/
+ ep93xx_stop_loop();
+
+#ifdef alsa_ac97_debug
+ ep93xx_dump_ac97_regs();
+#endif
+ DPRINTK("ep93xx_audio_init - EXIT\n");
+}
+
+/*====================================================================================*/
+
+
+static void print_audio_format( long format )
+{
+ switch( format ){
+ case SNDRV_PCM_FORMAT_S8:
+ DPRINTK( "AFMT_S8\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ DPRINTK( "AFMT_U8\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ DPRINTK( "AFMT_S16_LE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_BE:
+ DPRINTK( "AFMT_S16_BE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ DPRINTK( "AFMT_U16_LE\n" );
+ break;
+ case SNDRV_PCM_FORMAT_U16_BE:
+ DPRINTK( "AFMT_U16_BE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ DPRINTK( "AFMT_S24_LE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_BE:
+ DPRINTK( "AFMT_S24_BE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ DPRINTK( "AFMT_U24_LE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U24_BE:
+ DPRINTK( "AFMT_U24_BE\n" );
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ DPRINTK( "AFMT_S24_LE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_BE:
+ DPRINTK( "AFMT_S24_BE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U32_LE:
+ DPRINTK( "AFMT_U24_LE\n" );
+ break;
+
+ case SNDRV_PCM_FORMAT_U32_BE:
+ DPRINTK( "AFMT_U24_BE\n" );
+ break;
+ default:
+ DPRINTK( "ep93xx_i2s_Unsupported Audio Format\n" );
+ break;
+ }
+}
+
+static void audio_set_format( audio_stream_t * s, long val )
+{
+ DPRINTK( "ep93xx_i2s_audio_set_format enter. Format requested (%d) %d ",
+ (int)val,SNDRV_PCM_FORMAT_S16_LE);
+ print_audio_format( val );
+
+ switch( val ){
+ case SNDRV_PCM_FORMAT_S8:
+ s->audio_format = SNDRV_PCM_FORMAT_S8;
+ s->audio_stream_bitwidth = 8;
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ s->audio_format = SNDRV_PCM_FORMAT_U8;
+ s->audio_stream_bitwidth = 8;
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_S16_BE:
+ s->audio_format = SNDRV_PCM_FORMAT_S16_LE;
+ s->audio_stream_bitwidth = 16;
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ case SNDRV_PCM_FORMAT_U16_BE:
+ s->audio_format = SNDRV_PCM_FORMAT_U16_LE;
+ s->audio_stream_bitwidth = 16;
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_BE:
+ s->audio_format = SNDRV_PCM_FORMAT_S24_LE;
+ s->audio_stream_bitwidth = 24;
+ break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ case SNDRV_PCM_FORMAT_U24_BE:
+ s->audio_format = SNDRV_PCM_FORMAT_U24_LE;
+ s->audio_stream_bitwidth = 24;
+ break;
+
+ case SNDRV_PCM_FORMAT_U32_LE:
+ case SNDRV_PCM_FORMAT_U32_BE:
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S32_BE:
+ s->audio_format = SNDRV_PCM_FORMAT_S32_LE;
+ s->audio_stream_bitwidth = 32;
+ break;
+ default:
+ DPRINTK( "ep93xx_i2s_Unsupported Audio Format\n" );
+ break;
+ }
+
+ DPRINTK( "ep93xx_i2s_audio_set_format EXIT format set to be (%d) ", (int)s->audio_format );
+ print_audio_format( (long)s->audio_format );
+}
+
+static __inline__ unsigned long copy_to_user_S24_LE
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+
+ int total_to_count = to_count;
+ int *user_ptr = (int *)to; /* 32 bit user buffer */
+ int count;
+
+ count = 8 * stream->dma_num_channels;
+
+ while (to_count > 0){
+
+ __put_user( (int)( *dma_buffer_0++ ), user_ptr++ );
+ __put_user( (int)( *dma_buffer_0++ ), user_ptr++ );
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (int)( *dma_buffer_1++ ), user_ptr++ );
+ __put_user( (int)( *dma_buffer_1++ ), user_ptr++ );
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (int)( *dma_buffer_2++ ), user_ptr++ );
+ __put_user( (int)( *dma_buffer_2++ ), user_ptr++ );
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U24_LE
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+
+ int total_to_count = to_count;
+ unsigned int * user_ptr = (unsigned int *)to; /* 32 bit user buffer */
+ int count;
+
+ count = 8 * stream->dma_num_channels;
+
+ while (to_count > 0){
+ __put_user( ((unsigned int)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned int)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( ((unsigned int)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned int)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( ((unsigned int)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned int)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_S16_LE
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int total_to_count = to_count;
+ short * user_ptr = (short *)to; /* 16 bit user buffer */
+ int count;
+
+ count = 4 * stream->dma_num_channels;
+
+ while (to_count > 0){
+
+ __put_user( (short)( *dma_buffer_0++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_0++ ), user_ptr++ );
+
+ if( stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (short)( *dma_buffer_1++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_1++ ), user_ptr++ );
+ }
+
+ if( stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (short)( *dma_buffer_2++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_2++ ), user_ptr++ );
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U16_LE
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ short * user_ptr = (short *)to; /* 16 bit user buffer */
+
+ count = 4 * stream->dma_num_channels;
+
+ while (to_count > 0){
+
+ __put_user( ((unsigned short)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( ((unsigned short)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( ((unsigned short)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_S8
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ char * user_ptr = (char *)to; /* 8 bit user buffer */
+
+ count = 2 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (to_count > 0){
+
+ __put_user( (char)( *dma_buffer_0 ), user_ptr++ );
+ dma_buffer_0 += 4;
+ __put_user( (char)( *dma_buffer_0 ), user_ptr++ );
+ dma_buffer_0 += 4;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (char)( *dma_buffer_1 ), user_ptr++ );
+ dma_buffer_1 += 4;
+ __put_user( (char)( *dma_buffer_1 ), user_ptr++ );
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (char)( *dma_buffer_2 ), user_ptr++ );
+ dma_buffer_2 += 4;
+ __put_user( (char)( *dma_buffer_2 ), user_ptr++ );
+ dma_buffer_2 += 4;
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U8
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ char * user_ptr = (char *)to; /* 8 bit user buffer */
+
+ count = 2 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (to_count > 0){
+
+ __put_user( (char)( *dma_buffer_0 ) ^ 0x80, user_ptr++ );
+ dma_buffer_0 += 4;
+ __put_user( (char)( *dma_buffer_0 ) ^ 0x80, user_ptr++ );
+ dma_buffer_0 += 4;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (char)( *dma_buffer_1 ) ^ 0x80, user_ptr++ );
+ dma_buffer_1 += 4;
+ __put_user( (char)( *dma_buffer_1 ) ^ 0x80, user_ptr++ );
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (char)( *dma_buffer_2 ) ^ 0x80, user_ptr++ );
+ dma_buffer_2 += 4;
+ __put_user( (char)( *dma_buffer_2 ) ^ 0x80, user_ptr++ );
+ dma_buffer_2 += 4;
+ }
+ to_count -= count;
+ }
+ return total_to_count;
+}
+
+
+
+
+static __inline__ unsigned long copy_to_user_S16_LE_CM
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ short *dma_buffer_0 = (short *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int total_to_count = to_count;
+ short * user_ptr = (short *)to; /* 16 bit user buffer */
+ int count;
+
+
+ count = 4 * stream->dma_num_channels;
+
+ while (to_count > 0){
+ if(stream->audio_num_channels == 2){
+ __put_user( (short)( *dma_buffer_0++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_0++ ), user_ptr++ );
+ to_count -= count;
+ }
+ else{
+ dma_buffer_0++;
+ __put_user( (short)( *dma_buffer_0++ ), user_ptr++ );
+ to_count -= 2;
+ }
+
+ if( stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (short)( *dma_buffer_1++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_1++ ), user_ptr++ );
+ }
+
+ if( stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (short)( *dma_buffer_2++ ), user_ptr++ );
+ __put_user( (short)( *dma_buffer_2++ ), user_ptr++ );
+ }
+ //to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U16_LE_CM
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ unsigned short *dma_buffer_0 = (unsigned short *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ unsigned short * user_ptr = (unsigned short *)to; /* 16 bit user buffer */
+
+ count = 4 * stream->dma_num_channels;
+
+ while (to_count > 0){
+
+ if(stream->audio_num_channels == 2){
+ __put_user( ((unsigned short)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+ to_count -= count;
+ }
+ else{
+ dma_buffer_0++;
+ __put_user( ((unsigned short)( *dma_buffer_0++ )) ^ 0x8000, user_ptr++ );
+ to_count -= 2;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( ((unsigned short)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_1++ )) ^ 0x8000, user_ptr++ );
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( ((unsigned short)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ __put_user( ((unsigned short)( *dma_buffer_2++ )) ^ 0x8000, user_ptr++ );
+ }
+ //to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_S8_CM
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ unsigned short *dma_buffer_0 = (unsigned short *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ char * user_ptr = (char *)to; /* 8 bit user buffer */
+
+ count = 2 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (to_count > 0){
+ if(stream->audio_num_channels == 2){
+ __put_user( (char)( *dma_buffer_0++ >> 8), user_ptr++ );
+ //dma_buffer_0 += 4;
+ __put_user( (char)( *dma_buffer_0++ >> 8), user_ptr++ );
+ //dma_buffer_0 += 4;
+ to_count -= count;
+ }
+ else{
+ dma_buffer_0++ ;
+ __put_user( (char)( *dma_buffer_0++ >> 8), user_ptr++ );
+
+ to_count -= 1;
+ }
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (char)( *dma_buffer_1 ), user_ptr++ );
+ dma_buffer_1 += 4;
+ __put_user( (char)( *dma_buffer_1 ), user_ptr++ );
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (char)( *dma_buffer_2 ), user_ptr++ );
+ dma_buffer_2 += 4;
+ __put_user( (char)( *dma_buffer_2 ), user_ptr++ );
+ dma_buffer_2 += 4;
+ }
+ //to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U8_CM
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ unsigned short *dma_buffer_0 = (unsigned short *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ int total_to_count = to_count;
+ char * user_ptr = (char *)to; /* 8 bit user buffer */
+
+ count = 2 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (to_count > 0){
+ if(stream->audio_num_channels == 2){
+ __put_user( (char)( *dma_buffer_0++ >>8) ^ 0x80, user_ptr++ );
+ //dma_buffer_0 += 4;
+ __put_user( (char)( *dma_buffer_0++ >>8) ^ 0x80, user_ptr++ );
+ //dma_buffer_0 += 4;
+ to_count -= count;
+ }
+ else{
+ dma_buffer_0++;
+ __put_user( (char)( *dma_buffer_0++ >>8) ^ 0x80, user_ptr++ );
+ //dma_buffer_0 += 4;
+ to_count--;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __put_user( (char)( *dma_buffer_1 ) ^ 0x80, user_ptr++ );
+ dma_buffer_1 += 4;
+ __put_user( (char)( *dma_buffer_1 ) ^ 0x80, user_ptr++ );
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __put_user( (char)( *dma_buffer_2 ) ^ 0x80, user_ptr++ );
+ dma_buffer_2 += 4;
+ __put_user( (char)( *dma_buffer_2 ) ^ 0x80, user_ptr++ );
+ dma_buffer_2 += 4;
+ }
+ //to_count -= count;
+ }
+ return total_to_count;
+}
+
+static __inline__ unsigned long copy_to_user_U32
+(
+ audio_stream_t *stream,
+ const char *to,
+ unsigned long to_count
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+
+ if(__copy_to_user( (char *)to, dma_buffer_0, to_count))
+ {
+ return -EFAULT;
+ }
+ return to_count;
+}
+
+static __inline__ int copy_to_user_with_conversion
+(
+ audio_stream_t *stream,
+ const char *to,
+ int toCount,
+ int bCompactMode
+)
+{
+ int ret = 0;
+
+ if( toCount == 0 ){
+ DPRINTK("ep93xx_i2s_copy_to_user_with_conversion - nothing to copy!\n");
+ }
+
+ if( bCompactMode == 1 ){
+
+ switch( stream->audio_format ){
+
+ case SNDRV_PCM_FORMAT_S8:
+ ret = copy_to_user_S8_CM( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ ret = copy_to_user_U8_CM( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ ret = copy_to_user_S16_LE_CM( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ ret = copy_to_user_U16_LE_CM( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ //ret = copy_to_user_S24_LE( stream, to, toCount );
+ //break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ //ret = copy_to_user_U24_LE( stream, to, toCount );
+ //break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ default:
+ DPRINTK( "ep93xx_i2s copy to user unsupported audio format %x\n",stream->audio_format );
+ break;
+ }
+
+ }
+ else{
+
+ switch( stream->audio_format ){
+
+ case SNDRV_PCM_FORMAT_S8:
+ ret = copy_to_user_S8( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ ret = copy_to_user_U8( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ ret = copy_to_user_S16_LE( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ ret = copy_to_user_U16_LE( stream, to, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ //ret = copy_to_user_S24_LE( stream, to, toCount );
+ //break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ //ret = copy_to_user_U24_LE( stream, to, toCount );
+ //break;
+ DPRINTK( "ep93xx_i2s copy to user unsupported audio format %x\n",stream->audio_format );
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+
+ //__copy_to_user( (char *)to, from, toCount);
+ ret = copy_to_user_U32( stream, to, toCount );
+ break;
+ default:
+ DPRINTK( "ep93xx_i2s copy to user unsupported audio format\n" );
+ break;
+ }
+
+ }
+ return ret;
+}
+
+static __inline__ int copy_from_user_S24_LE
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+
+ unsigned int * user_buffer = (unsigned int *)from;
+ unsigned int data;
+
+ int toCount0 = toCount;
+ count = 8 * stream->dma_num_channels;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = (unsigned int)data;
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = (unsigned int)data;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = (unsigned int)data;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = (unsigned int)data;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = (unsigned int)data;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = (unsigned int)data;
+ }
+ toCount -= count;
+ }
+ return toCount0 / 2;
+}
+
+static __inline__ int copy_from_user_U24_LE
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+ unsigned int * user_buffer = (unsigned int *)from;
+ unsigned int data;
+
+ int toCount0 = toCount;
+ count = 8 * stream->dma_num_channels;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ }
+ toCount -= count;
+ }
+ return toCount0 / 2;
+}
+
+static __inline__ int copy_from_user_S16_LE
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ unsigned short *user_buffer = (unsigned short *)from;
+ unsigned short data;
+
+ int toCount0 = toCount;
+ int count;
+ count = 8 * stream->dma_num_channels;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = data;
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0++ = data;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = data;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = data;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = data;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = data;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 4;
+ }
+ return toCount0 / 2;
+}
+
+static __inline__ int copy_from_user_U16_LE
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+ unsigned short * user_buffer = (unsigned short *)from;
+ unsigned short data;
+
+ int toCount0 = toCount;
+ count = 8 * stream->dma_num_channels;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 4;
+ }
+ return toCount0 / 2;
+}
+
+static __inline__ int copy_from_user_S8
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ unsigned char * user_buffer = (unsigned char *)from;
+ unsigned char data;
+
+ int toCount0 = toCount;
+ count = 8 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (toCount > 0){
+ __get_user(data, user_buffer++);
+ *dma_buffer_0 = data;
+ dma_buffer_0 += 4;
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0 = data;
+ dma_buffer_0 += 4;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = data;
+ dma_buffer_1 += 4;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = data;
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = data;
+ dma_buffer_2 += 4;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = data;
+ dma_buffer_2 += 4;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 8;
+ }
+ return toCount0 / 4;
+}
+
+static __inline__ int copy_from_user_U8
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ unsigned char *user_buffer = (unsigned char *)from;
+ unsigned char data;
+
+ int toCount0 = toCount;
+ count = 8 * stream->dma_num_channels;
+
+ dma_buffer_0 ++;
+ dma_buffer_1 ++;
+ dma_buffer_2 ++;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0 = ((unsigned char)data ^ 0x80);
+ dma_buffer_0 += 4;
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0 = ((unsigned char)data ^ 0x80);
+ dma_buffer_0 += 4;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = ((unsigned char)data ^ 0x80);
+ dma_buffer_1 += 4;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = ((unsigned char)data ^ 0x80);
+ dma_buffer_1 += 4;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = ((unsigned char)data ^ 0x80);
+ dma_buffer_2 += 4;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = ((unsigned char)data ^ 0x80);
+ dma_buffer_2 += 4;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 8;
+ }
+ return toCount0 / 4;
+}
+
+static __inline__ int copy_from_user_S16_LE_CM
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ unsigned int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ unsigned int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ unsigned int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ unsigned short *user_buffer = (unsigned short *)from;
+ short data;
+ unsigned int val;
+ int toCount0 = toCount;
+ int count;
+ count = 4 * stream->dma_num_channels;
+
+ //printk("count=%x tocount\n",count,toCount);
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ //*dma_buffer_0++ = data;
+ val = (unsigned int)data & 0x0000ffff;
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0++ = ((unsigned int)data << 16) | val;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ //*dma_buffer_1++ = data;
+ val = (unsigned int)data & 0x0000ffff;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1++ = ((unsigned int)data << 16) | val;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ //*dma_buffer_2++ = data;
+ val = (unsigned int)data & 0x0000ffff;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2++ = ((unsigned int)data << 16) | val;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 /2 ;
+ }
+
+ return toCount0 ;
+}
+
+static __inline__ int copy_from_user_U16_LE_CM
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ int *dma_buffer_0 = (int *)stream->hwbuf[0];
+ int *dma_buffer_1 = (int *)stream->hwbuf[1];
+ int *dma_buffer_2 = (int *)stream->hwbuf[2];
+ int count;
+ unsigned short * user_buffer = (unsigned short *)from;
+ unsigned short data;
+ unsigned int val;
+ int toCount0 = toCount;
+ count = 4 * stream->dma_num_channels;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ //*dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+ val = (unsigned int)data & 0x0000ffff;
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ //*dma_buffer_0++ = ((unsigned int)data ^ 0x8000);
+ *dma_buffer_0++ = (((unsigned int)data << 16) | val) ^ 0x80008000;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ //*dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ val = (unsigned int)data & 0x0000ffff;
+ __get_user(data, user_buffer++);
+ //*dma_buffer_1++ = ((unsigned int)data ^ 0x8000);
+ *dma_buffer_1++ = (((unsigned int)data << 16) | val) ^ 0x80008000;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ //*dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ val = (unsigned int)data & 0x0000ffff;
+ __get_user(data, user_buffer++);
+ //*dma_buffer_2++ = ((unsigned int)data ^ 0x8000);
+ *dma_buffer_2++ = (((unsigned int)data << 16) | val) ^ 0x80008000;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0/2;
+ }
+ return toCount0 ;
+}
+
+static __inline__ int copy_from_user_S8_CM
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+ char *dma_buffer_1 = (char *)stream->hwbuf[1];
+ char *dma_buffer_2 = (char *)stream->hwbuf[2];
+ int count;
+ unsigned char * user_buffer = (unsigned char *)from;
+ unsigned char data;
+ int toCount0 = toCount;
+ count = 4 * stream->dma_num_channels;
+
+ dma_buffer_0++;
+ dma_buffer_1++;
+ dma_buffer_2++;
+
+ while (toCount > 0){
+ __get_user(data, user_buffer++);
+ *dma_buffer_0 = data;
+ *(dma_buffer_0 +1 ) = 0;
+ dma_buffer_0 += 2;
+
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0 = data;
+ *(dma_buffer_0 +1 ) = 0;
+ dma_buffer_0 += 2;
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = data;
+ dma_buffer_1 += 2;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = data;
+ dma_buffer_1 += 2;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = data;
+ dma_buffer_2 += 2;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = data;
+ dma_buffer_2 += 2;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 4;
+ }
+
+ return toCount0 / 2;
+}
+
+static __inline__ int copy_from_user_U8_CM
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ unsigned char *dma_buffer_0 = (unsigned char *)stream->hwbuf[0];
+ unsigned char *dma_buffer_1 = (unsigned char *)stream->hwbuf[1];
+ unsigned char *dma_buffer_2 = (unsigned char *)stream->hwbuf[2];
+ int count;
+ unsigned char *user_buffer = (unsigned char *)from;
+ unsigned char data;
+
+ int toCount0 = toCount;
+ count = 4 * stream->dma_num_channels;
+
+ dma_buffer_0 ++;
+ dma_buffer_1 ++;
+ dma_buffer_2 ++;
+
+ while (toCount > 0){
+
+ __get_user(data, user_buffer++);
+ *dma_buffer_0 = ((unsigned char)data ^ 0x80);
+ *(dma_buffer_0 +1 ) = 0;
+ dma_buffer_0 += 2;
+
+ if(stream->audio_num_channels == 2){
+ __get_user(data, user_buffer++);
+ }
+ *dma_buffer_0 = ((unsigned char)data ^ 0x80);
+ *(dma_buffer_0 +1 ) = 0;
+ dma_buffer_0 += 2;
+
+
+ if(stream->audio_channels_flag & CHANNEL_REAR ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = ((unsigned char)data ^ 0x80);
+ dma_buffer_1 += 2;
+ __get_user(data, user_buffer++);
+ *dma_buffer_1 = ((unsigned char)data ^ 0x80);
+ dma_buffer_1 += 2;
+ }
+
+ if(stream->audio_channels_flag & CHANNEL_CENTER_LFE ){
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = ((unsigned char)data ^ 0x80);
+ dma_buffer_2 += 2;
+ __get_user(data, user_buffer++);
+ *dma_buffer_2 = ((unsigned char)data ^ 0x80);
+ dma_buffer_2 += 2;
+ }
+ toCount -= count;
+ }
+
+ if(stream->audio_num_channels == 1){
+ return toCount0 / 4;
+ }
+
+ return toCount0 / 2;
+}
+
+static int copy_from_user_U32
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount
+)
+{
+ char *dma_buffer_0 = (char *)stream->hwbuf[0];
+
+ if (copy_from_user( (char *)dma_buffer_0, from, toCount))
+ {
+ return -EFAULT;
+ }
+
+ return toCount;
+
+}
+
+/*
+ * Returns negative for error
+ * Returns # of bytes transferred out of the from buffer
+ * for success.
+ */
+static __inline__ int copy_from_user_with_conversion
+(
+ audio_stream_t *stream,
+ const char *from,
+ int toCount,
+ int bCompactMode
+)
+{
+ int ret = 0;
+// DPRINTK("copy_from_user_with_conversion\n");
+ if( toCount == 0 ){
+ DPRINTK("ep93xx_i2s_copy_from_user_with_conversion - nothing to copy!\n");
+ }
+
+ if( bCompactMode == 1){
+
+ switch( stream->audio_format ){
+
+ case SNDRV_PCM_FORMAT_S8:
+ DPRINTK("SNDRV_PCM_FORMAT_S8 CM\n");
+ ret = copy_from_user_S8_CM( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ DPRINTK("SNDRV_PCM_FORMAT_U8 CM\n");
+ ret = copy_from_user_U8_CM( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S16_LE CM\n");
+ ret = copy_from_user_S16_LE_CM( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_U16_LE CM\n");
+ ret = copy_from_user_U16_LE_CM( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S24_LE CM\n");
+ //ret = copy_from_user_S24_LE( stream, from, toCount );
+ //break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_U24_LE CM\n");
+ //ret = copy_from_user_U24_LE( stream, from, toCount );
+ //break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S32_LE CM\n");
+ //break;
+ default:
+ DPRINTK( "ep93xx_i2s copy from user unsupported audio format\n" );
+ break;
+ }
+ }
+ else{
+ switch( stream->audio_format ){
+
+ case SNDRV_PCM_FORMAT_S8:
+ DPRINTK("SNDRV_PCM_FORMAT_S8\n");
+ ret = copy_from_user_S8( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U8:
+ DPRINTK("SNDRV_PCM_FORMAT_U8\n");
+ ret = copy_from_user_U8( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S16_LE\n");
+ ret = copy_from_user_S16_LE( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_U16_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_U16_LE\n");
+ ret = copy_from_user_U16_LE( stream, from, toCount );
+ break;
+
+ case SNDRV_PCM_FORMAT_S24_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S24_LE\n");
+ //ret = copy_from_user_S24_LE( stream, from, toCount );
+ //break;
+
+ case SNDRV_PCM_FORMAT_U24_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_U24_LE\n");
+ //ret = copy_from_user_U24_LE( stream, from, toCount );
+ //break;
+ DPRINTK( "ep93xx_i2s copy from user unsupported audio format\n" );
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ DPRINTK("SNDRV_PCM_FORMAT_S32_LE\n");
+ ret = copy_from_user_U32( stream, from, toCount );
+ break;
+ default:
+ DPRINTK( "ep93xx_i2s copy from user unsupported audio format\n" );
+ break;
+ }
+ }
+
+ return ret;
+}
+
+
+
+/*
+ * For audio playback, we convert samples of arbitrary format to be 32 bit
+ * for our hardware. We're scaling a user buffer to a dma buffer. So when
+ * report byte counts, we scale them acording to the ratio of DMA sample
+ * size to user buffer sample size. When we report # of DMA fragments,
+ * we don't scale that. So:
+ *
+ * Also adjust the size and number of dma fragments if sample size changed.
+ *
+ * Input format Input sample Output sample size ratio (out:in)
+ * bits channels size (bytes) CM non-CM CM non-CM
+ * 8 stereo 2 4 8 2:1 4:1
+ * 16 stereo 4 4 8 1:1 2:1
+ * 24 stereo 6 4 8 X 8:6 not a real case
+ *
+ */
+static void snd_ep93xx_dma2usr_ratio( audio_stream_t * stream,int bCompactMode )
+{
+ unsigned int dma_sample_size, user_sample_size;
+
+ if(bCompactMode == 1){
+ dma_sample_size = 4; /* each stereo sample is 2 * 32 bits */
+ }
+ else{
+ dma_sample_size = 8;
+ }
+
+ // If stereo 16 bit, user sample is 4 bytes.
+ // If stereo 8 bit, user sample is 2 bytes.
+ if(stream->audio_num_channels == 1){
+ user_sample_size = stream->audio_stream_bitwidth / 8;
+ }
+ else{
+ user_sample_size = stream->audio_stream_bitwidth / 4;
+ }
+
+ stream->dma2usr_ratio = dma_sample_size / user_sample_size;
+}
+
+/*---------------------------------------------------------------------------------------------*/
+
+static int snd_ep93xx_dma_free(struct snd_pcm_substream *substream ){
+
+
+ audio_state_t *state = substream->private_data;
+ audio_stream_t *stream = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_stream:state->input_stream;
+ int i;
+
+
+ DPRINTK("snd_ep93xx_dma_free - enter\n");
+ for( i = 0 ; i < stream->dma_num_channels ;i++ ){
+ ep93xx_dma_free( stream->dmahandles[i] );
+ }
+ DPRINTK("snd_ep93xx_dma_free - exit\n");
+ return 0;
+}
+
+static int snd_ep93xx_dma_config(struct snd_pcm_substream *substream ){
+
+ audio_state_t *state = substream->private_data;
+ audio_stream_t *stream = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_stream:state->input_stream;
+ int i,err = 0;
+
+ DPRINTK("snd_ep93xx_dma_config - enter\n");
+
+ for( i = 0 ; i < stream->dma_num_channels ;i++ ){
+
+ err = ep93xx_dma_request(&stream->dmahandles[i],
+ stream->devicename,
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_dma[i]:state->input_dma[i] );
+ if (err){
+ printk("snd_ep93xx_dma_config - exit ERROR dma request failed\n");
+ return err;
+ }
+ err = ep93xx_dma_config( stream->dmahandles[i],
+ IGNORE_CHANNEL_ERROR,
+ 0,
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ snd_ep93xx_dma_tx_callback:snd_ep93xx_dma_rx_callback,
+ (unsigned int)substream );
+ if (err){
+ printk("snd_ep93xx_dma_config - exit ERROR dma request failed\n");
+ return err;
+ }
+ }
+
+ DPRINTK("snd_ep93xx_dma_config - enter\n");
+ return err;
+}
+
+static void snd_ep93xx_dma_start( audio_state_t * state, audio_stream_t * stream )
+{
+ int err,i;
+
+ DPRINTK("snd_ep93xx_dma_start - enter\n");
+
+ for(i = 0 ;i < stream->dma_num_channels;i++)
+ err = ep93xx_dma_start( stream->dmahandles[i], 1,(unsigned int *) stream->dmahandles );
+
+ stream->active = 1;
+
+ DPRINTK("snd_ep93xx_dma_start - exit\n");
+}
+
+static void snd_ep93xx_dma_pause( audio_state_t * state, audio_stream_t * stream )
+{
+ int i;
+
+ DPRINTK("snd_ep93xx_dma_pause - enter\n");
+
+ for(i = 0 ;i < stream->dma_num_channels;i++)
+ ep93xx_dma_pause( stream->dmahandles[i], 1,(unsigned int *)stream->dmahandles );
+
+ stream->active = 0;
+ DPRINTK("snd_ep93xx_dma_pause - exit\n");
+
+}
+
+static void snd_ep93xx_dma_flush( audio_state_t * state, audio_stream_t * stream ){
+
+ int i;
+
+ DPRINTK("snd_ep93xx_dma_flush - enter\n");
+
+ for( i = 0 ; i < stream->dma_num_channels ; i++ )
+ ep93xx_dma_flush( stream->dmahandles[i] );
+
+ DPRINTK("snd_ep93xx_dma_flush - exit\n");
+}
+
+static void snd_ep93xx_deallocate_buffers( struct snd_pcm_substream *substream, audio_stream_t *stream )
+{
+ int i;
+ audio_channel_t *dma_chan;
+
+ DPRINTK("snd_ep93xx_deallocate_buffers - enter\n");
+
+ if( stream->dma_channels ){
+
+ for(i = 0;i < stream->dma_num_channels;i++){
+
+ dma_chan = &stream->dma_channels[i];
+
+ if( dma_chan->area ){
+
+ if( dma_chan->audio_buffers ){
+
+ kfree(dma_chan->audio_buffers);
+ dma_chan->audio_buffers = NULL;
+
+ }
+
+ kfree(dma_chan->area);
+ dma_chan->area = NULL;
+ }
+ }
+ kfree(stream->dma_channels);
+ stream->dma_channels = NULL;
+ }
+ DPRINTK("snd_ep93xx_deallocate_buffers - exit\n");
+}
+
+static int snd_ep93xx_allocate_buffers(struct snd_pcm_substream *substream, audio_stream_t *stream)
+{
+ audio_channel_t *channel;
+ unsigned int size,tmpsize,bufsize,bufextsize;
+ int i,j;
+
+
+ DPRINTK("snd_ep93xx_allocate_buffers - enter\n" );
+
+ if (stream->dma_channels){
+ printk("ep93xx_i2s %s BUSY\n",__FUNCTION__);
+ return -EBUSY;
+ }
+
+ stream->dma_channels = (audio_channel_t *)kmalloc(sizeof(audio_channel_t) * stream->dma_num_channels , GFP_KERNEL);
+
+ if (!stream->dma_channels){
+ printk(AUDIO_NAME ": unable to allocate dma_channels memory\n");
+ return - ENOMEM;
+ }
+
+ size = ( stream->dmasize / stream->dma_num_channels ) * stream->dma2usr_ratio;
+
+ for( i = 0; i < stream->dma_num_channels;i++){
+ channel = &stream->dma_channels[i];
+
+ channel->area = kmalloc( size, GFP_DMA );
+
+ if(!channel->area){
+ printk(AUDIO_NAME ": unable to allocate audio memory\n");
+ return -ENOMEM;
+ }
+ channel->bytes = size;
+ channel->addr = __virt_to_phys((int) channel->area);
+ memset( channel->area, 0, channel->bytes );
+
+ bufsize = ( stream->fragsize / stream->dma_num_channels ) * stream->dma2usr_ratio;
+ channel->audio_buff_count = size / bufsize;
+ bufextsize = size % bufsize;
+
+ if( bufextsize > 0 ){
+ channel->audio_buff_count++;
+ }
+
+ channel->audio_buffers = (audio_buf_t *)kmalloc(sizeof(audio_buf_t) * channel->audio_buff_count , GFP_KERNEL);
+
+ if (!channel->audio_buffers){
+ printk(AUDIO_NAME ": unable to allocate audio memory\n ");
+ return -ENOMEM;
+ }
+
+ tmpsize = size;
+
+ for( j = 0; j < channel->audio_buff_count; j++){
+
+ channel->audio_buffers[j].dma_addr = channel->addr + j * bufsize;
+
+ if( tmpsize >= bufsize ){
+ tmpsize -= bufsize;
+ channel->audio_buffers[j].bytes = bufsize;
+ channel->audio_buffers[j].reportedbytes = bufsize / stream->dma2usr_ratio;
+ }
+ else{
+ channel->audio_buffers[j].bytes = bufextsize;
+ channel->audio_buffers[j].reportedbytes = bufextsize / stream->dma2usr_ratio;
+ }
+ }
+ }
+
+ DPRINTK("snd_ep93xx_allocate_buffers -- exit SUCCESS\n" );
+ return 0;
+}
+
+/*
+ * DMA callback functions
+ */
+
+static void snd_ep93xx_dma_tx_callback
+(
+ ep93xx_dma_int_t DMAInt,
+ ep93xx_dma_dev_t device,
+ unsigned int user_data
+)
+{
+ int handle;
+ int i,chan;
+ unsigned int buf_id;
+
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)user_data;
+ audio_state_t *state = (audio_state_t *)(substream->private_data);
+ audio_stream_t *stream = state->output_stream;
+ audio_buf_t *buf;
+
+ switch( device )
+ {
+ case DMATx_I2S3:
+ DPRINTK( "snd_ep93xx_dma_tx_callback - DMATx_I2S3\n");
+ i = 2;
+ break;
+ case DMATx_I2S2:
+ DPRINTK( "snd_ep93xx_dma_tx_callback - DMATx_I2S2\n");
+ i = 1;
+ break;
+ case DMATx_I2S1:
+ default:
+ DPRINTK( "snd_ep93xx_dma_tx_callback - DMATx_I2S1\n");
+ i = 0;
+ break;
+ }
+
+ if(stream->audio_num_channels == 1){
+ chan = 0;
+ }
+ else{
+ chan = stream->audio_num_channels / 2 - 1;
+ }
+ handle = stream->dmahandles[i];
+
+ if(stream->stopped == 0){
+
+ if( ep93xx_dma_remove_buffer( handle, &buf_id ) >= 0 ){
+
+ buf = (audio_buf_t *)buf_id;
+ stream->bytecount += buf->reportedbytes;
+ ep93xx_dma_add_buffer( stream->dmahandles[i],
+ (unsigned int)buf->dma_addr,
+ 0,
+ buf->bytes,
+ 0,
+ (unsigned int) buf );
+ if(chan == i)
+ snd_pcm_period_elapsed(substream);
+ }
+ }
+}
+
+static void snd_ep93xx_dma_rx_callback
+(
+ ep93xx_dma_int_t DMAInt,
+ ep93xx_dma_dev_t device,
+ unsigned int user_data
+)
+{
+ int handle,i,chan;
+ unsigned int buf_id;
+ audio_buf_t *buf;
+
+ struct snd_pcm_substream *substream = (struct snd_pcm_substream *)user_data;
+ audio_state_t *state = (audio_state_t *)(substream->private_data);
+ audio_stream_t *stream = state->input_stream;
+
+ switch( device ){
+
+ case DMARx_I2S3:
+ DPRINTK( "snd_ep93xx_dma_rx_callback - DMARx_I2S3\n");
+ i = 2;
+ break;
+ case DMARx_I2S2:
+ DPRINTK( "snd_ep93xx_dma_rx_callback - DMARx_I2S2\n");
+ i = 1;
+ break;
+ case DMARx_I2S1:
+ default:
+ DPRINTK( "snd_ep93xx_dma_rx_callback - DMARx_I2S1\n");
+ i = 0;
+ break;
+ }
+
+ if(stream->audio_num_channels == 1){
+ chan = 0;
+ }
+ else{
+ chan = stream->audio_num_channels / 2 - 1;
+ }
+ handle = stream->dmahandles[i];
+
+ if( stream->stopped == 0 ){
+
+ if( ep93xx_dma_remove_buffer( handle, &buf_id ) >= 0 ){
+
+ buf = (audio_buf_t *)buf_id;
+ stream->bytecount += buf->reportedbytes;
+ ep93xx_dma_add_buffer( stream->dmahandles[i],
+ (unsigned int)buf->dma_addr,
+ 0,
+ buf->bytes,
+ 0,
+ (unsigned int) buf );
+ if( i == chan )
+ snd_pcm_period_elapsed(substream);
+ }
+ }
+}
+
+static int snd_ep93xx_release(struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *)substream->private_data;
+ audio_stream_t *stream = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_stream : state->input_stream;
+
+ DPRINTK("snd_ep93xx_release - enter\n");
+
+ down(&state->sem);
+ stream->active = 0;
+ stream->stopped = 0;
+ snd_ep93xx_deallocate_buffers(substream, stream);
+ up(&state->sem);
+
+ DPRINTK("snd_ep93xx_release - exit\n");
+
+ return 0;
+}
+
+static int ep93xx_ac97_pcm_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int r;
+ int iTempMasterVol,iTempHeadphoneVol,iTempMonoVol,iTempRecordSelect;
+ /*save the old mixer*/
+ iTempRecordSelect = peek(AC97_1A_RECORD_SELECT);
+ iTempMasterVol = peek( AC97_02_MASTER_VOL);
+ iTempHeadphoneVol = peek( AC97_04_HEADPHONE_VOL);
+ iTempMonoVol = peek( AC97_06_MONO_VOL);
+
+ runtime->hw.channels_min = 1;
+ runtime->hw.channels_max = 2;
+
+ ep93xx_audio_init();
+ /*ep93xx_init_ac97_controller();*/
+
+ /*reset the old output mixer*/
+ poke( AC97_02_MASTER_VOL, iTempMasterVol);
+ poke( AC97_04_HEADPHONE_VOL,iTempHeadphoneVol );
+ poke( AC97_06_MONO_VOL, iTempMonoVol);
+ poke( AC97_1A_RECORD_SELECT,iTempRecordSelect);
+
+ r = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ AC97_RATES_FRONT_DAC : AC97_RATES_ADC;
+
+ DPRINTK(" ep93xx_ac97_pcm_startup=%x\n",r);
+
+ return 0;
+}
+
+
+static int snd_ep93xx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ DPRINTK("snd_ep93xx_pcm_hw_params - enter\n");
+ return snd_pcm_lib_malloc_pages(substream,params_buffer_bytes(params));
+}
+
+static int snd_ep93xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+
+ DPRINTK("snd_ep93xx_pcm_hw_free - enter\n");
+ return snd_pcm_lib_free_pages(substream);
+}
+
+/*
+ *snd_ep93xx_pcm_prepare: need to finish these functions as lower
+ *chip_set_sample_format
+ *chip_set_sample_rate
+ *chip_set_channels
+ *chip_set_dma_setup
+ */
+
+static int snd_ep93xx_pcm_prepare_playback( struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *) substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->output_stream;
+
+ DPRINTK("snd_ep93xx_pcm_prepare_playback - enter\n");
+
+ ep93xx_audio_disable(1);
+ ep93xx_ac97_pcm_startup(substream);
+
+ snd_ep93xx_deallocate_buffers(substream,stream);
+
+ //if(runtime->channels % 2 != 0)
+ // return -1;
+
+ DPRINTK("The runtime item : \n");
+ DPRINTK("runtime->dma_addr = 0x%x\n", runtime->dma_addr);
+ DPRINTK("runtime->dma_area = 0x%x\n", runtime->dma_area);
+ DPRINTK("runtime->dma_bytes = %d\n", runtime->dma_bytes);
+ DPRINTK("runtime->frame_bits = %d\n", runtime->frame_bits);
+ DPRINTK("runtime->buffer_size = %d\n", runtime->buffer_size);
+ DPRINTK("runtime->period_size = %d\n", runtime->period_size);
+ DPRINTK("runtime->periods = %d\n", runtime->periods);
+ DPRINTK("runtime->rate = %d\n", runtime->rate);
+ DPRINTK("runtime->format = %d\n", runtime->format);
+ DPRINTK("runtime->channels = %d\n", runtime->channels);
+
+ /* set requestd format when available */
+ stream->audio_num_channels = runtime->channels;
+ if(stream->audio_num_channels == 1){
+ stream->dma_num_channels = 1;
+ }
+ else{
+ stream->dma_num_channels = runtime->channels / 2;
+ }
+
+ stream->audio_channels_flag = CHANNEL_FRONT;
+ if(stream->dma_num_channels == 2)
+ stream->audio_channels_flag |= CHANNEL_REAR;
+ if(stream->dma_num_channels == 3)
+ stream->audio_channels_flag |= CHANNEL_REAR | CHANNEL_CENTER_LFE;
+
+ stream->dmasize = runtime->dma_bytes;
+ stream->nbfrags = runtime->periods;
+ stream->fragsize = frames_to_bytes (runtime, runtime->period_size);
+ stream->bytecount = 0;
+
+ if( !state->codec_set_by_capture ){
+ state->codec_set_by_playback = 1;
+
+ if( stream->audio_rate != runtime->rate ){
+ ep93xx_set_samplerate( runtime->rate,0 );
+ }
+ //if( stream->audio_format != runtime->format ){
+ // snd_ep93xx_i2s_init((stream->audio_stream_bitwidth == 24));
+ //}
+ }
+ else{
+ audio_stream_t *s = state->input_stream;
+ if( runtime->format != s->audio_format)
+ return -1;
+ if( runtime->rate != s->audio_rate )
+ return -1;
+ }
+
+ stream->audio_format = runtime->format ;
+ ep93xx_set_hw_format(stream->audio_format,stream->audio_num_channels);
+
+
+ stream->audio_rate = runtime->rate;
+ audio_set_format( stream, runtime->format );
+ snd_ep93xx_dma2usr_ratio( stream,state->bCompactMode );
+
+ if( snd_ep93xx_allocate_buffers( substream, stream ) != 0 ){
+ snd_ep93xx_deallocate_buffers( substream, stream );
+ return -1;
+ }
+
+ ep93xx_audio_enable(1);
+
+ DPRINTK("snd_ep93xx_pcm_prepare_playback - exit\n");
+ return 0;
+}
+
+static int snd_ep93xx_pcm_prepare_capture( struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *) substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->input_stream;
+
+ ep93xx_audio_disable(0);
+ ep93xx_ac97_pcm_startup(substream);
+
+ snd_ep93xx_deallocate_buffers(substream,stream);
+
+ //if(runtime->channels % 2 != 0)
+ //return -1;
+
+ DPRINTK("snd_ep93xx_pcm_prepare_capture - enter\n");
+
+// printk("The runtime item : \n");
+// printk("runtime->dma_addr = 0x%x\n", runtime->dma_addr);
+// printk("runtime->dma_area = 0x%x\n", runtime->dma_area);
+// printk("runtime->dma_bytes = %d\n", runtime->dma_bytes);
+// printk("runtime->frame_bits = %d\n", runtime->frame_bits);
+// printk("runtime->buffer_size = %d\n", runtime->buffer_size);
+// printk("runtime->period_size = %d\n", runtime->period_size);
+// printk("runtime->periods = %d\n", runtime->periods);
+// printk("runtime->rate = %d\n", runtime->rate);
+// printk("runtime->format = %d\n", runtime->format);
+// printk("runtime->channels = %d\n", runtime->channels);
+
+ /* set requestd format when available */
+ stream->audio_num_channels = runtime->channels;
+ if(stream->audio_num_channels == 1){
+ stream->dma_num_channels = 1;
+ }
+ else{
+ stream->dma_num_channels = runtime->channels / 2;
+ }
+
+ stream->audio_channels_flag = CHANNEL_FRONT;
+ if(stream->dma_num_channels == 2)
+ stream->audio_channels_flag |= CHANNEL_REAR;
+ if(stream->dma_num_channels == 3)
+ stream->audio_channels_flag |= CHANNEL_REAR | CHANNEL_CENTER_LFE;
+
+ stream->dmasize = runtime->dma_bytes;
+ stream->nbfrags = runtime->periods;
+ stream->fragsize = frames_to_bytes (runtime, runtime->period_size);
+ stream->bytecount = 0;
+
+ if( !state->codec_set_by_playback ){
+ state->codec_set_by_capture = 1;
+
+ /*rate*/
+ if( stream->audio_rate != runtime->rate ){
+ ep93xx_set_samplerate( runtime->rate,1 );
+ }
+
+ /*mixer*/
+ ep93xx_set_recsource(SOUND_MASK_MIC|SOUND_MASK_LINE1 | SOUND_MASK_LINE);
+ poke( AC97_1C_RECORD_GAIN, 0);
+
+ /*format*/
+ //if( stream->audio_format != runtime->format ){
+ // snd_ep93xx_i2s_init((stream->audio_stream_bitwidth == 24));
+ //}
+ }
+ else{
+ audio_stream_t *s = state->output_stream;
+ if( runtime->format != s->audio_format)
+ return -1;
+ if( runtime->rate != s->audio_rate )
+ return -1;
+ }
+
+ stream->audio_format = runtime->format ;
+ ep93xx_set_hw_format(stream->audio_format,stream->audio_num_channels);
+
+
+ stream->audio_rate = runtime->rate;
+ audio_set_format( stream, runtime->format );
+ snd_ep93xx_dma2usr_ratio( stream,state->bCompactMode );
+
+ if( snd_ep93xx_allocate_buffers( substream, stream ) != 0 ){
+ snd_ep93xx_deallocate_buffers( substream, stream );
+ return -1;
+ }
+
+ ep93xx_audio_enable(0);
+
+ DPRINTK("snd_ep93xx_pcm_prepare_capture - exit\n");
+ return 0;
+}
+/*
+ *start/stop/pause dma translate
+ */
+static int snd_ep93xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ audio_state_t *state = (audio_state_t *)substream->private_data;
+ audio_stream_t *stream = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_stream:state->input_stream;
+ audio_buf_t *buf;
+ audio_channel_t *dma_channel;
+ int i,count,ret = 0;
+ unsigned long flags;
+
+ DPRINTK("snd_ep93xx_pcm_triger %d - enter \n",cmd);
+
+ switch (cmd){
+
+ case SNDRV_PCM_TRIGGER_START:
+
+ snd_ep93xx_dma_config( substream );
+
+ stream->stopped = 0;
+
+ if( !stream->active && !stream->stopped ){
+ stream->active = 1;
+ snd_ep93xx_dma_start( state, stream );
+ }
+
+ local_irq_save(flags);
+
+ for (i = 0; i < stream->dma_num_channels; i++){
+ dma_channel = &stream->dma_channels[i];
+
+ for(count = 0 ;count < dma_channel->audio_buff_count; count++){
+
+ buf = &dma_channel->audio_buffers[count];
+ ep93xx_dma_add_buffer( stream->dmahandles[i],
+ (unsigned int)buf->dma_addr,
+ 0,
+ buf->bytes,
+ 0,
+ (unsigned int) buf );
+ }
+ }
+
+ local_irq_restore(flags);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ stream->stopped = 1;
+ snd_ep93xx_dma_pause( state, stream );
+ snd_ep93xx_dma_flush( state, stream );
+ snd_ep93xx_dma_free( substream );
+ break;
+
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+ DPRINTK("snd_ep93xx_pcm_triger %d - exit \n",cmd);
+ return ret;
+}
+
+static snd_pcm_uframes_t snd_ep93xx_pcm_pointer_playback(struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *)(substream->private_data);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->output_stream;
+ snd_pcm_uframes_t pointer = 0;
+
+ pointer = bytes_to_frames( runtime,stream->bytecount );
+
+ if (pointer >= runtime->buffer_size){
+ pointer = 0;
+ stream->bytecount = 0;
+ }
+
+ DPRINTK("snd_ep93xx_pcm_pointer_playback - exit\n");
+ return pointer;
+}
+
+static snd_pcm_uframes_t snd_ep93xx_pcm_pointer_capture(struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *)(substream->private_data);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->input_stream;
+ snd_pcm_uframes_t pointer = 0;
+
+ pointer = bytes_to_frames( runtime,stream->bytecount );
+
+ if (pointer >= runtime->buffer_size){
+ pointer = 0;
+ stream->bytecount = 0;
+ }
+
+ DPRINTK("snd_ep93xx_pcm_pointer_capture - exit\n");
+ return pointer;
+}
+
+static int snd_ep93xx_pcm_open(struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ state->output_stream:state->input_stream;
+
+ DPRINTK("snd_ep93xx_pcm_open - enter\n");
+
+ down(&state->sem);
+
+ runtime->hw = ep93xx_ac97_pcm_hardware;
+
+ stream->dma_num_channels = AUDIO_DEFAULT_DMACHANNELS;
+ stream->dma_channels = NULL;
+ stream->audio_rate = 0;
+ stream->audio_stream_bitwidth = 0;
+
+ up(&state->sem);
+
+ DPRINTK("snd_ep93xx_pcm_open - exit\n");
+ return 0;
+}
+
+/*
+ *free the HW dma channel
+ *free the HW dma buffer
+ *free the Hw dma decrotion using function :kfree
+ */
+static int snd_ep93xx_pcm_close(struct snd_pcm_substream *substream)
+{
+ audio_state_t *state = (audio_state_t *)(substream->private_data);
+
+ DPRINTK("snd_ep93xx_pcm_close - enter\n");
+
+ snd_ep93xx_release(substream);
+
+ if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ state->codec_set_by_playback = 0;
+ else
+ state->codec_set_by_capture = 0;
+
+ DPRINTK("snd_ep93xx_pcm_close - exit\n");
+ return 0;
+}
+
+static int snd_ep93xx_pcm_copy_playback(struct snd_pcm_substream * substream,int channel,
+ snd_pcm_uframes_t pos,void __user *src, snd_pcm_uframes_t count)
+{
+
+ audio_state_t *state = (audio_state_t *)substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->output_stream ;
+ audio_channel_t *dma_channel;
+ int i;
+ int tocount = frames_to_bytes(runtime,count);
+
+ for( i = 0; i < stream->dma_num_channels; i++ ){
+
+ dma_channel = &stream->dma_channels[i];
+ stream->hwbuf[i] = dma_channel->area + ( frames_to_bytes(runtime,pos) * stream->dma2usr_ratio / stream->dma_num_channels );
+
+ }
+
+ if(copy_from_user_with_conversion(stream ,(const char*)src,(tocount * stream->dma2usr_ratio),state->bCompactMode) <=0 ){
+ DPRINTK(KERN_ERR "copy_from_user_with_conversion() failed\n");
+ return -EFAULT;
+ }
+
+ DPRINTK("snd_ep93xx_pcm_copy_playback - exit\n");
+ return 0;
+}
+
+
+static int snd_ep93xx_pcm_copy_capture(struct snd_pcm_substream * substream,int channel,
+ snd_pcm_uframes_t pos,void __user *src, snd_pcm_uframes_t count)
+{
+ audio_state_t *state = (audio_state_t *)substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ audio_stream_t *stream = state->input_stream ;
+ audio_channel_t *dma_channel;
+ int i;
+
+ int tocount = frames_to_bytes(runtime,count);
+
+ for( i = 0; i < stream->dma_num_channels; i++ ){
+
+ dma_channel = &stream->dma_channels[i];
+ stream->hwbuf[i] = dma_channel->area + ( frames_to_bytes(runtime,pos) * stream->dma2usr_ratio / stream->dma_num_channels );
+
+ }
+
+ if(copy_to_user_with_conversion(stream,(const char*)src,tocount,state->bCompactMode) <=0 ){
+
+ DPRINTK(KERN_ERR "copy_to_user_with_conversion() failed\n");
+ return -EFAULT;
+ }
+
+ DPRINTK("snd_ep93xx_pcm_copy_capture - exit\n");
+ return 0;
+}
+
+/*----------------------------------------------------------------------------------*/
+static unsigned short ep93xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ int val = -1;
+ /*volatile u32 *reg_addr;*/
+
+ DPRINTK(" number of codec:%x reg=%x\n",ac97->num,reg);
+ val=peek(reg);
+ if(val==-1){
+ printk(KERN_ERR "%s: read error (ac97_reg=%d )val=%x\n",
+ __FUNCTION__, reg, val);
+ return 0;
+ }
+
+ return val;
+}
+
+static void ep93xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ /*volatile u32 *reg_addr;*/
+ int ret;
+
+ DPRINTK(" number of codec:%x rge=%x val=%x\n",ac97->num,reg,val);
+ ret=poke(reg, val);
+ if(ret!=0){
+ printk(KERN_ERR "%s: write error (ac97_reg=%d val=%x)\n",
+ __FUNCTION__, reg, val);
+ }
+
+}
+
+static void ep93xx_ac97_reset(struct snd_ac97 *ac97)
+{
+
+ DPRINTK(" ep93xx_ac97_reset\n");
+ ep93xx_audio_init();
+
+}
+
+static struct snd_ac97_bus_ops ep93xx_ac97_ops = {
+ .read = ep93xx_ac97_read,
+ .write = ep93xx_ac97_write,
+ .reset = ep93xx_ac97_reset,
+};
+
+static struct snd_pcm *ep93xx_ac97_pcm;
+static struct snd_ac97 *ep93xx_ac97_ac97;
+
+static struct snd_pcm_ops snd_ep93xx_pcm_playback_ops = {
+ .open = snd_ep93xx_pcm_open,
+ .close = snd_ep93xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ep93xx_pcm_hw_params,
+ .hw_free = snd_ep93xx_pcm_hw_free,
+ .prepare = snd_ep93xx_pcm_prepare_playback,
+ .trigger = snd_ep93xx_pcm_trigger,
+ .pointer = snd_ep93xx_pcm_pointer_playback,
+ .copy = snd_ep93xx_pcm_copy_playback,
+
+};
+
+static struct snd_pcm_ops snd_ep93xx_pcm_capture_ops = {
+ .open = snd_ep93xx_pcm_open,
+ .close = snd_ep93xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ep93xx_pcm_hw_params,
+ .hw_free = snd_ep93xx_pcm_hw_free,
+ .prepare = snd_ep93xx_pcm_prepare_capture,
+ .trigger = snd_ep93xx_pcm_trigger,
+ .pointer = snd_ep93xx_pcm_pointer_capture,
+ .copy = snd_ep93xx_pcm_copy_capture,
+};
+
+/*--------------------------------------------------------------------------*/
+
+
+static int snd_ep93xx_pcm_new(struct snd_card *card, audio_state_t *state, struct snd_pcm **rpcm)
+{
+ struct snd_pcm *pcm;
+ int play = state->output_stream? 1 : 0;/*SNDRV_PCM_STREAM_PLAYBACK*/
+ int capt = state->input_stream ? 1 : 0;/*SNDRV_PCM_STREAM_CAPTURE*/
+ int ret = 0;
+
+ DPRINTK("snd_ep93xx_pcm_new - enter\n");
+
+ /* Register the new pcm device interface */
+ ret = snd_pcm_new(card, "EP93xx-AC97-PCM", 0, play, capt, &pcm);
+
+ if (ret){
+ DPRINTK("%s--%x:card=%x,play=%x,capt=%x,&pcm=%x\n",__FUNCTION__,ret,(int)card,play,capt,(int)pcm);
+ goto out;
+ }
+
+ /* allocate the pcm(DMA) memory */
+ ret = snd_pcm_lib_preallocate_pages_for_all(pcm, /*SNDRV_DMA_TYPE_DEV,0,*/SNDRV_DMA_TYPE_CONTINUOUS,snd_dma_continuous_data(GFP_KERNEL),128*1024,128*1024);
+
+ DPRINTK("The substream item : \n");
+ DPRINTK("pcm->streams[0].substream->dma_buffer.addr = 0x%x\n", pcm->streams[0].substream->dma_buffer.addr);
+ DPRINTK("pcm->streams[0].substream->dma_buffer.area = 0x%x\n", pcm->streams[0].substream->dma_buffer.area);
+ DPRINTK("pcm->streams[0].substream->dma_buffer.bytes = 0x%x\n", pcm->streams[0].substream->dma_buffer.bytes);
+ DPRINTK("pcm->streams[1].substream->dma_buffer.addr = 0x%x\n", pcm->streams[1].substream->dma_buffer.addr);
+ DPRINTK("pcm->streams[1].substream->dma_buffer.area = 0x%x\n", pcm->streams[1].substream->dma_buffer.area);
+ DPRINTK("pcm->streams[1].substream->dma_buffer.bytes = 0x%x\n", pcm->streams[1].substream->dma_buffer.bytes);
+
+ pcm->private_data = state;
+
+ /* seem to free the pcm data struct-->self dma buffer */
+ pcm->private_free = (void*) snd_pcm_lib_preallocate_free_for_all;
+
+ /* alsa pcm ops setting for SNDRV_PCM_STREAM_PLAYBACK */
+ if (play) {
+ int stream = SNDRV_PCM_STREAM_PLAYBACK;
+ snd_pcm_set_ops(pcm, stream, &snd_ep93xx_pcm_playback_ops);
+ }
+
+ /* alsa pcm ops setting for SNDRV_PCM_STREAM_CAPTURE */
+ if (capt) {
+ int stream = SNDRV_PCM_STREAM_CAPTURE;
+ snd_pcm_set_ops(pcm, stream, &snd_ep93xx_pcm_capture_ops);
+ }
+
+ if (rpcm)
+ *rpcm = pcm;
+ DPRINTK("snd_ep93xx_pcm_new - exit\n");
+out:
+ return ret;
+}
+
+#ifdef CONFIG_PM
+
+int ep93xx_ac97_do_suspend(struct snd_card *card, unsigned int state)
+{
+ if (card->power_state != SNDRV_CTL_POWER_D3cold) {
+ snd_pcm_suspend_all(ep93xx_ac97_pcm);
+ snd_ac97_suspend(ep93xx_ac97_ac97);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3cold);
+ }
+
+ return 0;
+}
+
+int ep93xx_ac97_do_resume(struct snd_card *card, unsigned int state)
+{
+ if (card->power_state != SNDRV_CTL_POWER_D0) {
+
+ snd_ac97_resume(ep93xx_ac97_ac97);
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ }
+
+ return 0;
+}
+
+int ep93xx_ac97_suspend(struct platform_device *_dev, u32 state, u32 level)
+{
+ struct snd_card *card = platform_get_drvdata(_dev);
+ int ret = 0;
+
+ if (card && level == SUSPEND_DISABLE)
+ ret = ep93xx_ac97_do_suspend(card, SNDRV_CTL_POWER_D3cold);
+
+ return ret;
+}
+
+int ep93xx_ac97_resume(struct platform_device *_dev, u32 level)
+{
+ struct snd_card *card = platform_get_drvdata(_dev);
+ int ret = 0;
+
+ if (card && level == RESUME_ENABLE)
+ ret = ep93xx_ac97_do_resume(card, SNDRV_CTL_POWER_D0);
+
+ return ret;
+}
+
+#else
+/*
+#define ep93xx_ac97_do_suspend NULL
+#define ep93xx_ac97_do_resume NULL
+#define ep93xx_ac97_suspend NULL
+#define ep93xx_ac97_resume NULL
+*/
+
+int ep93xx_ac97_do_suspend(struct snd_card *card, unsigned int state)
+{
+ return 0;
+}
+
+int ep93xx_ac97_do_resume(struct snd_card *card, unsigned int state)
+{
+ return 0;
+}
+
+int ep93xx_ac97_resume(struct platform_device *_dev, u32 level)
+{
+ struct snd_card *card = platform_get_drvdata(_dev);
+ int ret = 0;
+
+ //if (card && level == RESUME_ENABLE)
+ ret = ep93xx_ac97_do_resume(card, SNDRV_CTL_POWER_D0);
+
+ return ret;
+}
+
+int ep93xx_ac97_suspend(struct platform_device *_dev, u32 state, u32 level)
+{
+ struct snd_card *card = platform_get_drvdata(_dev);
+ int ret = 0;
+
+ //if (card && level == SUSPEND_DISABLE)
+ ret = ep93xx_ac97_do_suspend(card, SNDRV_CTL_POWER_D3cold);
+
+ return ret;
+}
+
+#endif
+
+
+
+/* module init & exit */
+static int __devinit ep93xx_ac97_probe(struct platform_device *dev)
+{
+ struct snd_card *card;
+ struct snd_ac97_bus *ac97_bus;
+ struct snd_ac97_template ac97_template;
+ int err = -ENOMEM;
+ struct resource *res = NULL;
+
+ DPRINTK("snd_ep93xx_probe - enter\n");
+
+ /* Enable audio early on, give the DAC time to come up. */
+ res = platform_get_resource( dev, IORESOURCE_MEM, 0);
+
+ if(!res) {
+ printk("error : platform_get_resource \n");
+ return -ENODEV;
+ }
+
+ if (!request_mem_region(res->start,res->end - res->start + 1, "snd-ac97-cs4202" )){
+ printk("error : request_mem_region\n");
+ return -EBUSY;
+ }
+
+ /*enable ac97 codec*/
+ ep93xx_audio_init();
+
+ /* register the soundcard */
+ card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+ THIS_MODULE, 0);
+ if (!card){
+ printk("AC97: snd_card_new error\n");
+ goto error;
+ }
+
+ card->dev = &dev->dev;
+ /*regist the new pcm device*/
+ err = snd_ep93xx_pcm_new(card, &audio_state, &ep93xx_ac97_pcm);
+ if (err){
+ printk("AC97: ep93xx_ac97_pcm_new error\n");
+ goto error;
+ }
+ if (card == NULL) {
+ DPRINTK(KERN_ERR "snd_card_new() failed\n");
+ goto error;
+ }
+
+ /*driver name*/
+ strcpy(card->driver, "CS4202A");
+ strcpy(card->shortname, "Cirrus Logic AC97 Audio ");
+ strcpy(card->longname, "Cirrus Logic AC97 Audio with CS4202A");
+
+ /*regist the new ac97 device*/
+ err = snd_ac97_bus(card, 0, &ep93xx_ac97_ops, NULL, &ac97_bus);
+ if (err){
+ printk("AC97: snd_ac97_bus error\n");
+ goto error;
+ }
+
+ memset(&ac97_template, 0, sizeof(ac97_template));
+ err = snd_ac97_mixer(ac97_bus, &ac97_template, &ep93xx_ac97_ac97);
+ if (err){
+ printk("AC97: snd_ac97_mixer error\n");
+ goto error;
+ }
+
+ /**/
+ ep93xx_audio_init();
+ /*setting the card device callback*/
+ //err = snd_card_set_pm_callback(card, ep93xx_ac97_do_suspend,ep93xx_ac97_do_resume, (void*)NULL);
+ //if(err != 0){
+ // printk("snd_card_set_pm_callback error\n");
+ //}
+
+ /*regist the new CARD device*/
+ err = snd_card_register(card);
+ if (err == 0) {
+ printk( KERN_INFO "Cirrus Logic ep93xx ac97 audio initialized\n" );
+ platform_set_drvdata(dev,card);
+ DPRINTK("snd_ep93xx_probe - exit\n");
+ return 0;
+ }
+
+error:
+ snd_card_free(card);
+ printk("snd_ep93xx_probe - error\n");
+ return err;
+
+return 0;
+}
+
+static int __devexit ep93xx_ac97_remove(struct platform_device *dev)
+{
+ struct resource *res;
+ struct snd_card *card = platform_get_drvdata(dev);
+
+ res = platform_get_resource( dev, IORESOURCE_MEM, 0);
+ release_mem_region(res->start, res->end - res->start + 1);
+
+ DPRINTK("snd_ep93xx_ac97_remove - enter\n");
+
+ if (card) {
+ snd_card_free(card);
+ platform_set_drvdata(dev, NULL);
+ }
+ DPRINTK("snd_ep93xx_remove - exit\n");
+
+return 0;
+}
+
+
+static struct platform_driver ep93xx_ac97_driver = {
+ .probe = ep93xx_ac97_probe,
+ .remove = __devexit_p (ep93xx_ac97_remove),
+ .suspend = ep93xx_ac97_suspend,
+ .resume = ep93xx_ac97_resume,
+ .driver = {
+ .name = "ep93xx-ac97",
+ },
+};
+
+
+static int __init ep93xx_ac97_init(void)
+{
+ int ret;
+
+ DPRINTK(KERN_INFO "%s: version %s\n", DRIVER_DESC, DRIVER_VERSION);
+ DPRINTK("snd_ep93xx_AC97_init - enter\n");
+ ret = platform_driver_register(&ep93xx_ac97_driver);
+ DPRINTK("snd_ep93xx_AC97_init - exit\n");
+ return ret;
+}
+
+static void __exit ep93xx_ac97_exit(void)
+{
+ DPRINTK("ep93xx_ac97_exit - enter\n");
+ return platform_driver_unregister(&ep93xx_ac97_driver);
+}
+
+module_init(ep93xx_ac97_init);
+module_exit(ep93xx_ac97_exit);
+
+MODULE_DESCRIPTION("Cirrus Logic audio module");
+MODULE_LICENSE("GPL");
--- /dev/null
+++ b/sound/arm/ep93xx-ac97.h
@@ -0,0 +1,89 @@
+/*
+ * linux/sound/arm/ep93xx-ac97.h -- ALSA PCM interface for the edb93xx ac97 audio
+ *
+ * Author: Fred Wei
+ * Created: July 19, 2005
+ * Copyright: Cirrus Logic, Inc.
+ *
+ * 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.
+ */
+
+#define EP93XX_DEFAULT_NUM_CHANNELS 2
+#define EP93XX_DEFAULT_FORMAT SNDRV_PCM_FORMAT_S16_LE
+#define EP93XX_DEFAULT_BIT_WIDTH 16
+#define MAX_DEVICE_NAME 20
+
+/*
+ * Buffer Management
+ */
+
+typedef struct {
+
+ unsigned char *area; /* virtual pointer */
+ dma_addr_t dma_addr; /* physical address */
+ size_t bytes;
+ size_t reportedbytes; /* buffer size */
+ int sent; /* indicates that dma has the buf */
+ char *start; /* points to actual buffer */
+
+} audio_buf_t;
+
+
+typedef struct {
+
+ unsigned char *area; /* virtual pointer */
+ dma_addr_t addr; /* physical address */
+ size_t bytes; /* buffer size in bytes */
+ unsigned char *buff_pos; /* virtual pointer */
+ audio_buf_t *audio_buffers; /* array of audio buffer structures */
+ int audio_buff_count;
+
+
+} audio_channel_t;
+
+typedef struct audio_stream_s {
+
+ /* dma stuff */
+ int dmahandles[3]; /* handles for dma driver instances */
+ char devicename[MAX_DEVICE_NAME]; /* string - name of device */
+ int dma_num_channels; /* 1, 2, or 3 DMA channels */
+ audio_channel_t *dma_channels;
+ u_int nbfrags; /* nbr of fragments i.e. buffers */
+ u_int fragsize; /* fragment i.e. buffer size */
+ u_int dmasize;
+ int bytecount; /* nbr of processed bytes */
+ int externedbytecount; /* nbr of processed bytes */
+ volatile int active; /* actually in progress */
+ volatile int stopped; /* might be active but stopped */
+ char *hwbuf[3];
+ long audio_rate;
+ long audio_num_channels; /* Range: 1 to 6 */
+ int audio_channels_flag;
+ long audio_format;
+ long audio_stream_bitwidth; /* Range: 8, 16, 24 */
+ int dma2usr_ratio;
+
+} audio_stream_t;
+
+
+/*
+ * State structure for one instance
+ */
+typedef struct {
+
+ audio_stream_t *output_stream;
+ audio_stream_t *input_stream;
+ ep93xx_dma_dev_t output_dma[3];
+ ep93xx_dma_dev_t input_dma[3];
+ char *output_id[3];
+ char *input_id[3];
+ struct semaphore sem; /* to protect against races in attach() */
+ int codec_set_by_playback;
+ int codec_set_by_capture;
+ int DAC_bit_width; /* 16, 20, 24 bits */
+ int bCompactMode; /* set if 32bits = a stereo sample */
+
+} audio_state_t;
+
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