diff options
| author | florian <florian@3c298f89-4303-0410-b956-a3cf2f4a3e73> | 2009-11-25 23:43:48 +0000 |
|---|---|---|
| committer | florian <florian@3c298f89-4303-0410-b956-a3cf2f4a3e73> | 2009-11-25 23:43:48 +0000 |
| commit | 288fe380457a72eaf1a6b6201326e5a51b82976c (patch) | |
| tree | 4e11bba416827d9797e6dac0e98bafd030a6ef0f /target/linux/ep93xx/patches-2.6.30/010-ep93xx-snd-ac97.patch | |
| parent | bd245df66fd2d2474807fcfb636119dd782f8da4 (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.patch | 3808 |
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; ++ |