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authorflorian <florian@3c298f89-4303-0410-b956-a3cf2f4a3e73>2009-11-25 23:43:48 +0000
committerflorian <florian@3c298f89-4303-0410-b956-a3cf2f4a3e73>2009-11-25 23:43:48 +0000
commit288fe380457a72eaf1a6b6201326e5a51b82976c (patch)
tree4e11bba416827d9797e6dac0e98bafd030a6ef0f /target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch
parentbd245df66fd2d2474807fcfb636119dd782f8da4 (diff)
[ep93xx] add support for the Simplemachines Sim.One board
git-svn-id: svn://svn.openwrt.org/openwrt/trunk@18540 3c298f89-4303-0410-b956-a3cf2f4a3e73
Diffstat (limited to 'target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch')
-rw-r--r--target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch3808
1 files changed, 3808 insertions, 0 deletions
diff --git a/target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch b/target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch
new file mode 100644
index 0000000000..36f316be17
--- /dev/null
+++ b/target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch
@@ -0,0 +1,3808 @@
+--- 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;
++