* [PATCH] audigyls driver.
@ 2004-07-02 16:42 James Courtier-Dutton
2004-07-02 17:05 ` Takashi Iwai
0 siblings, 1 reply; 4+ messages in thread
From: James Courtier-Dutton @ 2004-07-02 16:42 UTC (permalink / raw)
To: alsa-devel
[-- Attachment #1: Type: text/plain, Size: 178 bytes --]
Here is a patch against the current CVS.
I have implemented a separate DMA buffer for the period table list.
I have implemented a list for the remove/rename ctls.
Cheers
James
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Index: alsa-driver/pci/emu10k1/audigyls.c
===================================================================
RCS file: /cvsroot/alsa/alsa-driver/pci/emu10k1/audigyls.c,v
retrieving revision 1.5
diff -u -r1.5 audigyls.c
--- alsa-driver/pci/emu10k1/audigyls.c 1 Jul 2004 08:38:28 -0000 1.5
+++ alsa-driver/pci/emu10k1/audigyls.c 2 Jul 2004 16:39:00 -0000
@@ -1,6 +1,7 @@
/*
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver AUDIGYLS chips
+ * Version: 0.8
*
* FEATURES currently supported:
* Front, Rear and Center/LFE.
@@ -17,16 +18,41 @@
* So, to record from the MIC, set the MIC Playback volume to max,
* unmute the MIC and turn up the MASTER Playback volume.
* So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume.
- * The Digital/Analog switch must be in Analog mode for CAPTURE.
+ *
+ * The only playback controls that currently do anything are: -
+ * Analog Front
+ * Analog Rear
+ * Analog Center/LFE
+ * SPDIF Front
+ * SPDIF Rear
+ * SPDIF Center/LFE
+ *
+ * For capture from Mic in or Line in.
+ * (The AudigyLS uses the AC97 playback channel for capture. The AC97 capture channel is not used at all.)
+ * Master
+ * Mic (The one marked as playback)
+ * Mic boost
+ * Line (The one marked as playback)
+ * Digital/Analog ( switch must be in Analog mode for CAPTURE. )
+ * CAPTURE feedback into PLAYBACK
+ *
+ * Changelog:
+ * Support interrupts per period.
+ * Removed noise from Center/LFE channel when in Analog mode.
+ * Rename and remove mixer controls.
+ * 0.0.6
+ * Use separate card based DMA buffer for periods table list.
+ * 0.0.7
+ * Change remove and rename ctrls into lists.
+ * 0.0.8
+ * Try to fix capture sources.
*
* BUGS:
* Some stability problems when unloading the snd-audigyls kernel module.
* --
*
* TODO:
- * Need to add a way to select capture source.
* 4 Capture channels, only one implemented so far.
- * Need to find out what the AC97 chip actually does.
* Other rates apart from 48khz not implemented.
* MIDI
* --
@@ -107,6 +133,7 @@
#define INTE_CH_0_HALF_LOOP 0x00000100 /* Channel 0 half loop */
#define INTE_TIMER 0x00000008 /* Interrupt based on Timer */
+#define UNKNOWN14 0x10 /* Unknown ??. Defaults to 0 */
#define HCFG 0x14 /* Hardware config register */
#define HCFG_LOCKSOUNDCACHE 0x00000008 /* 1 = Cancel bustmaster accesses to soundcache */
@@ -114,19 +141,28 @@
#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
/* Should be set to 1 when the EMU10K1 is */
/* completely initialized. */
+#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */
+#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
+
+#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
/********************************************************************************************************/
/* Audigy LS pointer-offset register set, accessed through the PTR and DATA registers */
/********************************************************************************************************/
/* Initally all registers from 0x00 to 0x3f have zero contents. */
-#define PLAYBACK_UNKNOWN0 0x00 /* Something used in playback */
-#define PLAYBACK_UNKNOWN1 0x01 /* Something used in playback */
-#define PLAYBACK_UNKNOWN2 0x02 /* Something used in playback */
-#define PLAYBACK_UNKNOWN3 0x03 /* Something ?? */
+#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
+ /* One list entry: 4 bytes for DMA address,
+ * 4 bytes for period_size << 16.
+ * One list entry is 8 bytes long.
+ * One list entry for each period in the buffer.
+ */
+#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
+#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
+#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
-#define PLAYBACK_BUFFER_SIZE 0x05 /* Playback buffer size. win2000 uses 0x04000000 */
-#define PLAYBACK_POINTER 0x06 /* Playback buffer pointer. Sample currently in DAC */
+#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
+#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
#define PLAYBACK_UNKNOWN7 0x07 /* Something used in playback */
#define PLAYBACK_UNKNOWN8 0x08 /* Something used in playback */
#define PLAYBACK_UNKNOWN9 0x09 /* Something ?? */
@@ -134,10 +170,7 @@
#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
#define CAPTURE_UNKNOWN13 0x13 /* Something used in capture */
-#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
-
-#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
-#define PLAYBACk_LAST_SAMPLE 0x20 /* The sample currently being played */
+#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */
/* 0x21 - 0x3f unused */
#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
/* Playback (0x1<<channel_id) */
@@ -189,6 +222,12 @@
/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
* But as the jack is shared, use 0xf00.
* The Windows2000 driver uses 0x0000000f for both digital and analog.
+ * 0xf00 introduces interesting noises onto the Center/LFE.
+ * If you turn the volume up, you hear computer noise,
+ * e.g. mouse moving, changing between app windows etc.
+ * So, I am going to set this to 0x0000000f all the time now,
+ * same as the windows driver does.
+ * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
*/
#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
@@ -242,6 +281,10 @@
#define PCM_REAR_CHANNEL 1
#define PCM_CENTER_LFE_CHANNEL 2
#define PCM_UNKNOWN_CHANNEL 3
+#define CONTROL_FRONT_CHANNEL 0
+#define CONTROL_REAR_CHANNEL 3
+#define CONTROL_CENTER_LFE_CHANNEL 1
+#define CONTROL_UNKNOWN_CHANNEL 2
typedef struct snd_audigyls_voice audigyls_voice_t;
typedef struct snd_audigyls_channel audigyls_channel_t;
@@ -297,7 +340,8 @@
audigyls_channel_t channels[4];
audigyls_channel_t capture_channel;
u32 spdif_bits[4]; /* s/pdif out setup */
- int digital_analog;
+ int spdif_enable;
+ int capture_source;
struct snd_dma_device dma_dev;
struct snd_dma_buffer buffer;
@@ -316,8 +360,8 @@
.rates = SNDRV_PCM_RATE_48000,
.rate_min = 48000,
.rate_max = 48000,
- .channels_min = 1,
- .channels_max = 6,
+ .channels_min = 2, //1,
+ .channels_max = 2, //6,
.buffer_bytes_max = (32*1024),
.period_bytes_min = 64,
.period_bytes_max = (16*1024),
@@ -641,17 +685,27 @@
audigyls_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
audigyls_pcm_t *epcm = runtime->private_data;
- int voice = epcm->voice->number;
- voice=epcm->channel_id;
+ int voice = voice=epcm->channel_id;
+ u32 *table_base = (u32 *)(emu->buffer.area+(8*16*voice));
+ u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
+ int i;
- //printk("prepare:voice_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",voice, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
- snd_audigyls_ptr_write(emu, 0x00, voice, 0);
- snd_audigyls_ptr_write(emu, 0x01, voice, 0);
- snd_audigyls_ptr_write(emu, 0x02, voice, 0);
+ //snd_printk("prepare:voice_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",voice, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ for(i=0; i < runtime->periods; i++) {
+ table_base[i*2]=runtime->dma_addr+(i*period_size_bytes);
+ table_base[(i*2)+1]=period_size_bytes<<16;
+ }
+
+ snd_audigyls_ptr_write(emu, PLAYBACK_LIST_ADDR, voice, emu->buffer.addr+(8*16*voice));
+ snd_audigyls_ptr_write(emu, PLAYBACK_LIST_SIZE, voice, (runtime->periods - 1) << 19);
+ snd_audigyls_ptr_write(emu, PLAYBACK_LIST_PTR, voice, 0);
snd_audigyls_ptr_write(emu, PLAYBACK_DMA_ADDR, voice, runtime->dma_addr);
- snd_audigyls_ptr_write(emu, PLAYBACK_BUFFER_SIZE, voice, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+ snd_audigyls_ptr_write(emu, PLAYBACK_PERIOD_SIZE, voice, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
snd_audigyls_ptr_write(emu, PLAYBACK_POINTER, voice, 0);
- snd_audigyls_ptr_write(emu, 0x07, voice, 0);
+ snd_audigyls_ptr_write(emu, 0x07, voice, 0x0);
snd_audigyls_ptr_write(emu, 0x08, voice, 0);
snd_audigyls_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
#if 0
@@ -683,7 +737,6 @@
snd_audigyls_ptr_write(emu, CAPTURE_DMA_ADDR, voice, runtime->dma_addr);
snd_audigyls_ptr_write(emu, CAPTURE_BUFFER_SIZE, voice, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
snd_audigyls_ptr_write(emu, CAPTURE_POINTER, voice, 0);
- snd_audigyls_ptr_write(emu, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC or Line in */
snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_audigyls_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<voice));
return 0;
@@ -752,15 +805,19 @@
audigyls_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
audigyls_pcm_t *epcm = runtime->private_data;
- snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0;
int channel = epcm->voice->number;
channel=epcm->channel_id;
if (!epcm->running)
return 0;
+ ptr3 = snd_audigyls_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
ptr1 = snd_audigyls_ptr_read(emu, PLAYBACK_POINTER, channel);
+ ptr4 = snd_audigyls_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
+ if (ptr3 != ptr4) ptr1 = snd_audigyls_ptr_read(emu, PLAYBACK_POINTER, channel);
ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr2+= (ptr4 >> 3) * runtime->period_size;
ptr=ptr2;
if (ptr >= runtime->buffer_size)
ptr -= runtime->buffer_size;
@@ -901,7 +958,8 @@
snd_audigyls_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
udelay(1000);
// disable audio
- outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
+ //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
+ outl(0, chip->port + HCFG);
/* FIXME: We need to stop and DMA transfers here.
* But as I am not sure how yet, we cannot from the dma pages.
* So we can fix: snd-malloc: Memory leak? pages not freed = 8
@@ -953,7 +1011,7 @@
return IRQ_NONE;
//printk(KERN_INFO "interrupt status = %08x, chip=%p, channel=%p\n", status,chip, chip->channels);
- stat76 = snd_audigyls_ptr_read(chip, 0x76, 0);
+ stat76 = snd_audigyls_ptr_read(chip, EXTENDED_INT, 0);
//mask = IPR_CH_0_LOOP|IPR_CH_0_HALF_LOOP;
mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
for(i = 0; i < 4; i++) {
@@ -977,7 +1035,7 @@
}
}
- snd_audigyls_ptr_write(chip, 0x76, 0, stat76);
+ snd_audigyls_ptr_write(chip, EXTENDED_INT, 0, stat76);
spin_lock(&chip->emu_lock);
// acknowledge the interrupt if necessary
outl(status, chip->port+IPR);
@@ -1037,7 +1095,7 @@
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
- 32*1024, 32*1024)) < 0)
+ 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
return err;
}
@@ -1090,8 +1148,8 @@
spin_lock_init(&chip->voice_lock);
chip->port = pci_resource_start(pci, 0);
- if ((chip->res_port = request_region(chip->port, 8,
- "My Chip")) == NULL) {
+ if ((chip->res_port = request_region(chip->port, 0x20,
+ "snd_audigyls")) == NULL) {
snd_audigyls_free(chip);
printk(KERN_ERR "cannot allocate the port\n");
return -EBUSY;
@@ -1109,8 +1167,8 @@
memset(&chip->dma_dev, 0, sizeof(chip->dma_dev));
chip->dma_dev.type = SNDRV_DMA_TYPE_DEV;
chip->dma_dev.dev = snd_dma_pci_data(pci);
-
- if(snd_dma_alloc_pages(&chip->dma_dev, 32 * 1024, &chip->buffer) < 0) {
+ /* This stores the periods table. */
+ if(snd_dma_alloc_pages(&chip->dma_dev, 1024, &chip->buffer) < 0) {
snd_audigyls_free(chip);
return -ENOMEM;
}
@@ -1188,12 +1246,13 @@
//snd_audigyls_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
/* Analog or Digital output */
snd_audigyls_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
- snd_audigyls_ptr_write(chip, SPDIF_SELECT2, 0, 0x0b000000); /* 0x000b0000 for analog, from win2000 drivers */
- chip->digital_analog = 0; /* Set digital SPDIF output on */
+ snd_audigyls_ptr_write(chip, SPDIF_SELECT2, 0, 0x000b0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers */
+ chip->spdif_enable = 0; /* Set digital SPDIF output off */
+ chip->capture_source = 3; /* Set CAPTURE_SOURCE */
//snd_audigyls_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
//snd_audigyls_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
- snd_audigyls_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c80000); /* goes to 0x40c81000 when not doing SPDIF IN/OUT */
+ snd_audigyls_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
snd_audigyls_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
snd_audigyls_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
snd_audigyls_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
@@ -1204,16 +1263,20 @@
for(ch = 0; ch < 4; ch++) {
snd_audigyls_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
snd_audigyls_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
- snd_audigyls_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x30303030);
- snd_audigyls_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x30303030);
+ snd_audigyls_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
+ snd_audigyls_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
}
snd_audigyls_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
+ chip->capture_source = 3; /* Set CAPTURE_SOURCE */
- outl(0, chip->port+0x18);
- snd_audigyls_intr_enable(chip, 0x105);
-
- outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
+ //outl(0, chip->port+GPIO);
+ outl(0x005f03a3, chip->port+GPIO); /* Analog */
+ //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */
+ snd_audigyls_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
+ //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
+ outl(0x00001409, chip->port+HCFG);
+
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
snd_audigyls_free(chip);
@@ -1345,14 +1408,10 @@
static int snd_audigyls_shared_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
- static char *texts[2] = { "Digital", "Analog " };
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item > 1)
- uinfo->value.enumerated.item = 1;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
return 0;
}
@@ -1361,7 +1420,7 @@
{
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
- ucontrol->value.enumerated.item[0] = emu->digital_analog;
+ ucontrol->value.enumerated.item[0] = emu->spdif_enable;
return 0;
}
@@ -1371,23 +1430,29 @@
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
+ u32 mask;
val = ucontrol->value.enumerated.item[0] ;
- change = (emu->digital_analog != val);
+ change = (emu->spdif_enable != val);
if (change) {
- emu->digital_analog = val;
- if (val == 0) {
+ emu->spdif_enable = val;
+ if (val == 1) {
/* Digital */
snd_audigyls_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_audigyls_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
snd_audigyls_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_audigyls_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
+ mask = inl(emu->port + GPIO) & ~0x101;
+ outl(mask, emu->port + GPIO);
+
} else {
/* Analog */
- snd_audigyls_ptr_write(emu, SPDIF_SELECT1, 0, 0xf00);
+ snd_audigyls_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_audigyls_ptr_write(emu, SPDIF_SELECT2, 0, 0x000b0000);
snd_audigyls_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_audigyls_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
+ mask = inl(emu->port + GPIO) | 0x101;
+ outl(mask, emu->port + GPIO);
}
}
return change;
@@ -1396,12 +1461,63 @@
static snd_kcontrol_new_t snd_audigyls_shared_spdif __devinitdata =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Analog/Digital Output Jack",
+ .name = "SPDIF Out",
.info = snd_audigyls_shared_spdif_info,
.get = snd_audigyls_shared_spdif_get,
.put = snd_audigyls_shared_spdif_put
};
+static int snd_audigyls_capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[4] = { "SPDIF ", "What U Hear", "Unknown ", "AC97 " };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_audigyls_capture_source_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ audigyls_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_source;
+ return 0;
+}
+
+static int snd_audigyls_capture_source_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ audigyls_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+ u32 source;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->capture_source != val);
+ if (change) {
+ emu->capture_source = val;
+ source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+ mask = snd_audigyls_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
+ snd_audigyls_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_audigyls_capture_source __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = snd_audigyls_capture_source_info,
+ .get = snd_audigyls_capture_source_get,
+ .put = snd_audigyls_capture_source_put
+};
+
static int snd_audigyls_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
@@ -1472,7 +1588,7 @@
.put = snd_audigyls_spdif_put
};
-static int snd_audigyls_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo, int channel_id)
+static int snd_audigyls_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
@@ -1480,31 +1596,6 @@
uinfo->value.integer.max = 255;
return 0;
}
-static int snd_audigyls_volume_info_front(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
-{
- int channel_id = PCM_FRONT_CHANNEL;
- return snd_audigyls_volume_info(kcontrol, uinfo, channel_id);
-}
-static int snd_audigyls_volume_info_center_lfe(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
-{
- int channel_id = PCM_CENTER_LFE_CHANNEL;
- return snd_audigyls_volume_info(kcontrol, uinfo, channel_id);
-}
-static int snd_audigyls_volume_info_unknown(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
-{
- int channel_id = PCM_UNKNOWN_CHANNEL;
- return snd_audigyls_volume_info(kcontrol, uinfo, channel_id);
-}
-static int snd_audigyls_volume_info_rear(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
-{
- int channel_id = PCM_REAR_CHANNEL;
- return snd_audigyls_volume_info(kcontrol, uinfo, channel_id);
-}
-static int snd_audigyls_volume_info_feedback(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
-{
- int channel_id = 1;
- return snd_audigyls_volume_info(kcontrol, uinfo, channel_id);
-}
static int snd_audigyls_volume_get(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
@@ -1518,35 +1609,65 @@
return 0;
}
-static int snd_audigyls_volume_get_front(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_get_spdif_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_FRONT_CHANNEL;
+ int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_get_center_lfe(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_get_spdif_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_CENTER_LFE_CHANNEL;
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_get_unknown(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_get_spdif_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_UNKNOWN_CHANNEL;
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_get_rear(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_get_spdif_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_REAR_CHANNEL;
+ int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
}
+static int snd_audigyls_volume_get_analog_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
+static int snd_audigyls_volume_get_analog_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_audigyls_volume_get_analog_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_audigyls_volume_get_analog_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
+}
+
static int snd_audigyls_volume_get_feedback(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
@@ -1567,34 +1688,63 @@
snd_audigyls_ptr_write(emu, reg, channel_id, value);
return 1;
}
-static int snd_audigyls_volume_put_front(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_put_spdif_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_FRONT_CHANNEL;
+ int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_put_center_lfe(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_put_spdif_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_CENTER_LFE_CHANNEL;
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_put_unknown(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_put_spdif_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_UNKNOWN_CHANNEL;
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
-static int snd_audigyls_volume_put_rear(snd_kcontrol_t * kcontrol,
+static int snd_audigyls_volume_put_spdif_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
- int channel_id = PCM_REAR_CHANNEL;
+ int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
+static int snd_audigyls_volume_put_analog_front(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_FRONT_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_audigyls_volume_put_analog_center_lfe(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_CENTER_LFE_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_audigyls_volume_put_analog_unknown(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_UNKNOWN_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+static int snd_audigyls_volume_put_analog_rear(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ int channel_id = CONTROL_REAR_CHANNEL;
+ int reg = PLAYBACK_VOLUME2;
+ return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
+}
+
static int snd_audigyls_volume_put_feedback(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
@@ -1603,43 +1753,76 @@
return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
-static snd_kcontrol_new_t snd_audigyls_volume_control_front =
+static snd_kcontrol_new_t snd_audigyls_volume_control_analog_front =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Analog Front Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_analog_front,
+ .put = snd_audigyls_volume_put_analog_front
+};
+static snd_kcontrol_new_t snd_audigyls_volume_control_analog_center_lfe =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Front Volume",
- .info = snd_audigyls_volume_info_front,
- .get = snd_audigyls_volume_get_front,
- .put = snd_audigyls_volume_put_front
+ .name = "Analog Center/LFE Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_analog_center_lfe,
+ .put = snd_audigyls_volume_put_analog_center_lfe
};
-static snd_kcontrol_new_t snd_audigyls_volume_control_center_lfe =
+static snd_kcontrol_new_t snd_audigyls_volume_control_analog_unknown =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Center/LFE Volume",
- .info = snd_audigyls_volume_info_center_lfe,
- .get = snd_audigyls_volume_get_center_lfe,
- .put = snd_audigyls_volume_put_center_lfe
+ .name = "Analog Unknown Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_analog_unknown,
+ .put = snd_audigyls_volume_put_analog_unknown
};
-static snd_kcontrol_new_t snd_audigyls_volume_control_unknown =
+static snd_kcontrol_new_t snd_audigyls_volume_control_analog_rear =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Unknown Volume",
- .info = snd_audigyls_volume_info_unknown,
- .get = snd_audigyls_volume_get_unknown,
- .put = snd_audigyls_volume_put_unknown
+ .name = "Analog Rear Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_analog_rear,
+ .put = snd_audigyls_volume_put_analog_rear
};
-static snd_kcontrol_new_t snd_audigyls_volume_control_rear =
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_front =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "PCM Rear Volume",
- .info = snd_audigyls_volume_info_rear,
- .get = snd_audigyls_volume_get_rear,
- .put = snd_audigyls_volume_put_rear
+ .name = "SPDIF Front Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_spdif_front,
+ .put = snd_audigyls_volume_put_spdif_front
};
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_center_lfe =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Center/LFE Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_spdif_center_lfe,
+ .put = snd_audigyls_volume_put_spdif_center_lfe
+};
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_unknown =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Unknown Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_spdif_unknown,
+ .put = snd_audigyls_volume_put_spdif_unknown
+};
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_rear =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SPDIF Rear Volume",
+ .info = snd_audigyls_volume_info,
+ .get = snd_audigyls_volume_get_spdif_rear,
+ .put = snd_audigyls_volume_put_spdif_rear
+};
+
static snd_kcontrol_new_t snd_audigyls_volume_control_feedback =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "CAPTURE feedback into PLAYBACK",
- .info = snd_audigyls_volume_info_feedback,
+ .info = snd_audigyls_volume_info,
.get = snd_audigyls_volume_get_feedback,
.put = snd_audigyls_volume_put_feedback
};
@@ -1664,7 +1847,6 @@
return snd_ctl_find_id(card, &sid);
}
-#if 0
static int rename_ctl(snd_card_t *card, const char *src, const char *dst)
{
snd_kcontrol_t *kctl = ctl_find(card, src);
@@ -1674,26 +1856,86 @@
}
return -ENOENT;
}
-#endif
static int __devinit snd_audigyls_mixer(audigyls_t *emu)
{
int err;
snd_kcontrol_t *kctl;
snd_card_t *card = emu->card;
- if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_front, emu)) == NULL)
+ char **c;
+ static char *audigyls_remove_ctls[] = {
+ "Master Mono Playback Switch",
+ "Master Mono Playback Volume",
+ "3D Control - Switch",
+ "3D Control Sigmatel - Depth",
+ "PCM Playback Switch",
+ "PCM Playback Volume",
+ "CD Playback Switch",
+ "CD Playback Volume",
+ "Phone Playback Switch",
+ "Phone Playback Volume",
+ "Video Playback Switch",
+ "Video Playback Volume",
+ "PC Speaker Playback Switch",
+ "PC Speaker Playback Volume",
+ "Mono Output Select",
+ "Capture Source",
+ "Capture Switch",
+ "Capture Volume",
+ "External Amplifier",
+ "Sigmatel 4-Speaker Stereo Playback Switch",
+ "Sigmatel Surround Phase Inversion Playback ",
+ NULL
+ };
+ static char *audigyls_rename_ctls[] = {
+ "Master Playback Switch", "Capture Switch",
+ "Master Playback Volume", "Capture Volume",
+ "Line Playback Switch", "AC97 Line Capture Switch",
+ "Line Playback Volume", "AC97 Line Capture Volume",
+ "Aux Playback Switch", "AC97 Aux Capture Switch",
+ "Aux Playback Volume", "AC97 Aux Capture Volume",
+ "Mic Playback Switch", "AC97 Mic Capture Switch",
+ "Mic Playback Volume", "AC97 Mic Capture Volume",
+ "Mic Select", "AC97 Mic Select",
+ "Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)",
+ NULL
+ };
+#if 1
+ for (c=audigyls_remove_ctls; *c; c++)
+ remove_ctl(card, *c);
+ for (c=audigyls_rename_ctls; *c; c += 2)
+ rename_ctl(card, c[0], c[1]);
+#endif
+
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_analog_front, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_analog_rear, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_analog_center_lfe, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_analog_unknown, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_spdif_front, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
- if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_rear, emu)) == NULL)
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_spdif_rear, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
- if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_center_lfe, emu)) == NULL)
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_spdif_center_lfe, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
- if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_unknown, emu)) == NULL)
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_spdif_unknown, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
@@ -1709,6 +1951,10 @@
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_capture_source, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
if ((kctl = ctl_find(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT))) != NULL) {
/* already defined by ac97, remove it */
/* FIXME: or do we need both controls? */
^ permalink raw reply [flat|nested] 4+ messages in thread* Re: [PATCH] audigyls driver.
2004-07-02 16:42 [PATCH] audigyls driver James Courtier-Dutton
@ 2004-07-02 17:05 ` Takashi Iwai
2004-07-05 13:03 ` Pate plugin problem? Francisco Moraes
0 siblings, 1 reply; 4+ messages in thread
From: Takashi Iwai @ 2004-07-02 17:05 UTC (permalink / raw)
To: James Courtier-Dutton; +Cc: alsa-devel
At Fri, 02 Jul 2004 17:42:09 +0100,
James Courtier-Dutton wrote:
>
> Here is a patch against the current CVS.
>
> I have implemented a separate DMA buffer for the period table list.
> I have implemented a list for the remove/rename ctls.
Thanks, I applied the patch to cvs.
The only changes I made are:
- remove space letters from capture source enum strings
- remove obsolete magic definitions
Takashi
-------------------------------------------------------
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^ permalink raw reply [flat|nested] 4+ messages in thread
* Pate plugin problem?
2004-07-02 17:05 ` Takashi Iwai
@ 2004-07-05 13:03 ` Francisco Moraes
2004-07-05 13:15 ` Takashi Iwai
0 siblings, 1 reply; 4+ messages in thread
From: Francisco Moraes @ 2004-07-05 13:03 UTC (permalink / raw)
To: alsa-devel
On the EMU10K1X driver, when I capture sound at 48000, everything sounds
good. Whenever I use a lower sample rate, the sound has noticeble cracks.
It seems this is a problem in the rate conversion plugin. Has anyone
tested this kind of downward rate conversion sucessfully? Any
suggestions on how to find/fix the problem?
Francisco
-------------------------------------------------------
This SF.Net email sponsored by Black Hat Briefings & Training.
Attend Black Hat Briefings & Training, Las Vegas July 24-29 -
digital self defense, top technical experts, no vendor pitches,
unmatched networking opportunities. Visit www.blackhat.com
^ permalink raw reply [flat|nested] 4+ messages in thread
* Re: Pate plugin problem?
2004-07-05 13:03 ` Pate plugin problem? Francisco Moraes
@ 2004-07-05 13:15 ` Takashi Iwai
0 siblings, 0 replies; 4+ messages in thread
From: Takashi Iwai @ 2004-07-05 13:15 UTC (permalink / raw)
To: Francisco Moraes; +Cc: alsa-devel
At Mon, 05 Jul 2004 09:03:54 -0400,
Francisco Moraes wrote:
>
> On the EMU10K1X driver, when I capture sound at 48000, everything sounds
> good. Whenever I use a lower sample rate, the sound has noticeble cracks.
>
> It seems this is a problem in the rate conversion plugin. Has anyone
> tested this kind of downward rate conversion sucessfully? Any
> suggestions on how to find/fix the problem?
It works fine on ICH and VIA82xx chips.
A possible reason would be that the period size isn't aligned suitably
for your soundchip when the rate conversion is applied.
Check the hw_params in the case of non-48kHz.
Takashi
-------------------------------------------------------
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Attend Black Hat Briefings & Training, Las Vegas July 24-29 -
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^ permalink raw reply [flat|nested] 4+ messages in thread
end of thread, other threads:[~2004-07-05 13:15 UTC | newest]
Thread overview: 4+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2004-07-02 16:42 [PATCH] audigyls driver James Courtier-Dutton
2004-07-02 17:05 ` Takashi Iwai
2004-07-05 13:03 ` Pate plugin problem? Francisco Moraes
2004-07-05 13:15 ` Takashi Iwai
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