* [PATCH] AudigyLS updates.
@ 2004-06-25 21:37 James Courtier-Dutton
2004-06-28 10:48 ` Takashi Iwai
0 siblings, 1 reply; 6+ messages in thread
From: James Courtier-Dutton @ 2004-06-25 21:37 UTC (permalink / raw)
To: ALSA development
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AudigyLS now supports Digital(AC3)/Analog outputs. 5.1 Surround
PLAYBACK. MIC and LINE IN CAPTURE.
The MIC and LINE IN use the AC97 chip.
The AC97 uses the PLAYBACK(Not CAPTURE) controls in alsamixer for
controlling CAPTURE. Then, instead of sending that output to the
speakers, it sends it to the Philips 1361T ADC. This results in high
quality ADC conversion.
E.g. For LINE IN capture, set LINE IN playback volume and unmute it.
Set MASTER volume.
There is a new control for "CAPTURE feedback into PLAYBACK". If one does
not want any feedback, leave it at it's minimum.
Once I have figured out all the controls in the AC97 and what they do
for the AudigyLS, I will rename them so that they make sense to the user.
Please see attached patch files against the current CVS.
Still TODO:
1) Set multiple sample rates.
2) Add a CAPUTRE source switch for switch between "MIC", "Line in",
"SPDIF in", "TAD in", "AUX in" and "What u hear". Currently, one cannot
switch to capture "SPDIF in" or "What u hear", but all the other inputs
are possible via switches in the AC97.
3) Set multiple periods per buffer.
4) MIDI in/out.
5) Cleaner module unload.
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 1.2
diff -u -r1.2 audigyls.c
--- alsa-driver/pci/emu10k1/audigyls.c 23 Jun 2004 17:21:30 -0000 1.2
+++ alsa-driver/pci/emu10k1/audigyls.c 25 Jun 2004 20:15:46 -0000
@@ -5,12 +5,22 @@
* FEATURES currently supported:
* Front, Rear and Center/LFE.
* Surround40 and Surround51.
- * Capture from MIC input.
+ * Capture from MIC an LINE IN input.
* SPDIF digital playback of PCM stereo and AC3/DTS works.
- * (One can use a standard stereo mini-jack to two RCA plugs cable.
+ * (One can use a standard mono mini-jack to one RCA plugs cable.
+ * or one can use a standard stereo mini-jack to two RCA plugs cable.
* Plug one of the RCA plugs into the Coax input of the external decoder/receiver.)
+ * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. )
+ * Notes on how to capture sound:
+ * The AC97 is used in the PLAYBACK direction.
+ * The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC.
+ * 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.
*
* BUGS:
+ * Some stability problems when unloading the snd-audigyls kernel module.
* --
*
* TODO:
@@ -45,6 +55,7 @@
*
*/
#include <sound/driver.h>
+#include <linux/delay.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
@@ -98,6 +109,7 @@
#define INTE 0x0c /* Interrupt enable register */
#define INTE_CH_0_LOOP 0x00000800 /* Channel 0 loop */
#define INTE_CH_0_HALF_LOOP 0x00000100 /* Channel 0 half loop */
+#define INTE_TIMER 0x00000008 /* Interrupt based on Timer */
#define HCFG 0x14 /* Hardware config register */
@@ -111,25 +123,29 @@
/* 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_DMA_ADDR 0x04 /* Something used in playback */
-#define PLAYBACK_BUFFER_SIZE 0x05 /* Something used in playback */
+#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_UNKNOWN7 0x07 /* Something used in playback */
#define PLAYBACK_UNKNOWN8 0x08 /* Something used in playback */
#define PLAYBACK_UNKNOWN9 0x09 /* Something ?? */
-#define CAPTURE_DMA_ADDR 0x10 /* Something used in capture */
-#define CAPTURE_BUFFER_SIZE 0x11 /* Something used in capture */
+#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
+#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 BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
+#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) */
+ /* Capture (0x100<<channel_id) */
/* The Digital out jack is shared with the Center/LFE Analogue output.
* The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
* For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
@@ -145,8 +161,6 @@
/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
* A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs.
*/
-
-
#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */
#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */
#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */
@@ -175,34 +189,63 @@
#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
-#define SPDIF_SELECT 0x45 /* Enables SPDIF or Analogue outputs 0-Analogue, 0xf00-SPDIF */
+#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
+ /* 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.
+ */
#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
-#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - MIC input. */
-#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - ??, 2-??, 3-?? */
-#define CAPTURE_SOURCE_CHANNEL3 0x000f0000
-#define CAPTURE_SOURCE_LOOPBACK 0x0000ffff
+#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF input. */
+#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear. 2 - ?? */
+#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */
+#define CAPTURE_SOURCE_LOOPBACK 0x0000ffff /* Default 0x00e4 */
#define CAPTURE_VOLUME1 0x61 /* Capture volume per voice */
#define CAPTURE_VOLUME2 0x62 /* Capture volume per voice */
-#define UNKNOWN_VOLUME 0x66 /* Unknown volume per voice */
-#define PLAYBACK_VOLUME 0x6a /* Playback volume per voice */
+
+#define PLAYBACK_ROUTING1 0x63 /* Playback routing. Effects AC3 output. Default 0x32765410 */
+#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
+#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
+#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
+#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
+
+#define PLAYBACK_ROUTING2 0x64 /* Unknown Routing. Effects AC3 output. Default 0x76767676 */
+#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */
+#define PLAYBACK_VOLUME1 0x66 /* Playback volume per voice. Set to the same PLAYBACK_VOLUME(0x6a) */
+ /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
+#define CAPTURE_ROUTING1 0x67 /* Playback Routing. Default 0x32765410 */
+#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */
+#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
+#define PLAYBACK_VOLUME2 0x6a /* Playback volume per voice. Does not effect AC3 output */
+#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
#define UART1 0x6c /* Uart, used in setting sample rates, bits per sample etc. */
#define UART2 0x6d /* Uart, used in setting sample rates, bits per sample etc. */
-#define ROUTING1 0x71 /* Some sort of routing. default = 0xf0000000 */
-#define ROUTING2 0x72 /* Some sort of routing. default = 0x0f0f003f */
-#define CHIP_VERSION 0x74 /* P17 Chip version */
+#define UNKNOWN_UART1 0x6e /* Uart, Unknown. */
+#define UNKNOWN_UART2 0x6f /* Uart, Unknown. */
+#define UNKNOWN70 0x70 /* Unknown. Readonly. Default 00108000 00108000 00500000 00500000 */
+#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
+ /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
+ /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
+#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
+#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */
+#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */
+#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */
+#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */
+#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */
#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */
/* Sets which Interrupts are enabled. */
#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */
/* Shows which interrupts are active at the moment. */
-#define EXTENDED_INT_TIMER 0x79 /* Used by both playback and capture interrupt handler */
+#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */
+#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */
+#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */
/* Causes interrupts based on timer intervals. */
#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
#define PCM_FRONT_CHANNEL 0
-#define PCM_CENTER_LFE_CHANNEL 1
-#define PCM_UNKNOWN_CHANNEL 2
-#define PCM_REAR_CHANNEL 3
+#define PCM_REAR_CHANNEL 1
+#define PCM_CENTER_LFE_CHANNEL 2
+#define PCM_UNKNOWN_CHANNEL 3
#define chip_t audigyls_t
@@ -260,6 +303,7 @@
audigyls_channel_t channels[4];
audigyls_channel_t capture_channel;
u32 spdif_bits[4]; /* s/pdif out setup */
+ int digital_analog;
struct snd_dma_device dma_dev;
struct snd_dma_buffer buffer;
@@ -402,8 +446,9 @@
{
audigyls_pcm_t *epcm = snd_magic_cast(audigyls_pcm_t, runtime->private_data, return);
- if (epcm)
+ if (epcm) {
snd_magic_kfree(epcm);
+ }
}
static void snd_audigyls_pcm_interrupt(audigyls_t *emu, audigyls_voice_t *voice)
@@ -439,6 +484,7 @@
snd_pcm_runtime_t *runtime = substream->runtime;
epcm = snd_magic_kcalloc(audigyls_pcm_t, 0, GFP_KERNEL);
+
if (epcm == NULL)
return -ENOMEM;
epcm->emu = chip;
@@ -603,6 +649,7 @@
audigyls_pcm_t *epcm = snd_magic_cast(audigyls_pcm_t, runtime->private_data, return -ENXIO);
int voice = epcm->voice->number;
voice=epcm->channel_id;
+
//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);
@@ -612,18 +659,19 @@
snd_audigyls_ptr_write(emu, PLAYBACK_POINTER, voice, 0);
snd_audigyls_ptr_write(emu, 0x07, voice, 0);
snd_audigyls_ptr_write(emu, 0x08, voice, 0);
- snd_audigyls_ptr_write(emu, 0x67, 0x0, 0x76543210); /* FIXME: What is this */
+ snd_audigyls_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
#if 0
snd_audigyls_ptr_write(emu, SPCS0, 0,
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
- 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT | SPCS_NOTAUDIODATA);
+ 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
+ }
#endif
- snd_audigyls_ptr_write(emu, SPCS0, 0, 0x2108006);
- //snd_audigyls_ptr_write(emu, 0x75, 0, 0x33); /* Routing of some sort */
- //snd_audigyls_ptr_write(emu, 0x75, 0, 0x11 << voice); /* Routing of some sort */
- snd_audigyls_ptr_write(emu, 0x75, 0, snd_audigyls_ptr_read(emu, 0x75, 0) | (0x11<<voice));
+ //snd_audigyls_ptr_write(emu, SPCS0, 0, 0x2108006);
+ //snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, 0x33); /* Routing of some sort */
+ //snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, 0x11 << voice); /* Routing of some sort */
+ snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_audigyls_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x11<<voice));
return 0;
}
@@ -641,8 +689,8 @@
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, 0x60, 0x0, 0x0); /* FIXME: What is this */
- snd_audigyls_ptr_write(emu, 0x75, 0, snd_audigyls_ptr_read(emu, 0x75, 0) | (0x110000<<voice));
+ 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;
}
@@ -660,12 +708,12 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- snd_audigyls_ptr_write(emu, 0x40, 0, snd_audigyls_ptr_read(emu, 0x40, 0)|(0x1<<channel));
+ snd_audigyls_ptr_write(emu, BASIC_INTERRUPT, 0, snd_audigyls_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x1<<channel));
epcm->running = 1;
break;
case SNDRV_PCM_TRIGGER_STOP:
- snd_audigyls_ptr_write(emu, 0x40, 0, snd_audigyls_ptr_read(emu, 0x40, 0) & ~(0x1<<channel));
- snd_audigyls_ptr_write(emu, 0x75, 0, snd_audigyls_ptr_read(emu, 0x75, 0) & ~(0x11<<channel));
+ snd_audigyls_ptr_write(emu, BASIC_INTERRUPT, 0, snd_audigyls_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x1<<channel));
+ snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_audigyls_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x11<<channel));
epcm->running = 0;
break;
default:
@@ -688,12 +736,12 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
- snd_audigyls_ptr_write(emu, 0x40, 0, snd_audigyls_ptr_read(emu, 0x40, 0)|(0x100<<channel));
+ snd_audigyls_ptr_write(emu, BASIC_INTERRUPT, 0, snd_audigyls_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
epcm->running = 1;
break;
case SNDRV_PCM_TRIGGER_STOP:
- snd_audigyls_ptr_write(emu, 0x40, 0, snd_audigyls_ptr_read(emu, 0x40, 0) & ~(0x100<<channel));
- snd_audigyls_ptr_write(emu, 0x75, 0, snd_audigyls_ptr_read(emu, 0x75, 0) & ~(0x110000<<channel));
+ snd_audigyls_ptr_write(emu, BASIC_INTERRUPT, 0, snd_audigyls_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+ snd_audigyls_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_audigyls_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
epcm->running = 0;
break;
default:
@@ -852,11 +900,24 @@
static int snd_audigyls_free(audigyls_t *chip)
{
- snd_audigyls_ptr_write(chip, 0x40, 0, 0);
- // disable interrupts
- outl(0, chip->port + INTE);
- // disable audio
- outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
+ if (chip->res_port != NULL) { /* avoid access to already used hardware */
+ // disable interrupts
+ snd_audigyls_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
+ outl(0, chip->port + INTE);
+ snd_audigyls_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
+ udelay(1000);
+ // disable audio
+ outl(HCFG_LOCKSOUNDCACHE, 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
+ */
+ }
+ // release the data
+#if 1
+ if (chip->buffer.area)
+ snd_dma_free_pages(&chip->dma_dev, &chip->buffer);
+#endif
// release the i/o port
if (chip->res_port) {
@@ -866,7 +927,6 @@
// release the irq
if (chip->irq >= 0)
free_irq(chip->irq, (void *)chip);
- // release the data
snd_magic_kfree(chip);
return 0;
}
@@ -963,13 +1023,13 @@
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_audigyls_capture_ops);
break;
case 1:
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_center_lfe_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_rear_ops);
break;
case 2:
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_unknown_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_center_lfe_ops);
break;
case 3:
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_rear_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_audigyls_playback_unknown_ops);
break;
}
@@ -980,22 +1040,23 @@
for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
substream;
- substream = substream->next)
+ substream = substream->next) {
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
32*1024, 32*1024)) < 0)
return err;
-
+ }
for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
substream;
- substream = substream->next)
- if ((err = snd_pcm_lib_preallocate_pages(substream,
+ substream = substream->next) {
+ if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
64*1024, 64*1024)) < 0)
- return err;
+ return err;
+ }
if (rpcm)
*rpcm = pcm;
@@ -1066,8 +1127,10 @@
pci_read_config_byte(pci, PCI_REVISION_ID, (char *)&chip->revision);
pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
+#if 0
printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model,
chip->revision, chip->serial);
+#endif
outl(0, chip->port + INTE);
@@ -1116,9 +1179,8 @@
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
- /* Select analogue output */
- //snd_audigyls_ptr_write(chip, 0x41, 0, 0x70f); // ???
- //snd_audigyls_ptr_write(chip, 0x65, 0, 0x1000);
+ snd_audigyls_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
+ snd_audigyls_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
/* Write 0x8000 to AC97_REC_GAIN to mute it. */
outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
@@ -1130,16 +1192,30 @@
snd_audigyls_ptr_write(chip, 0x44, 0, 0x2108006);
#endif
- snd_audigyls_ptr_write(chip, 0x72, 0, 0xf0f003f);
-
- snd_audigyls_ptr_write(chip, 0x71, 0, 0xf0000000);
- //snd_audigyls_ptr_write(chip, 0x61, 0, 0x0);
- //snd_audigyls_ptr_write(chip, 0x62, 0, 0x0);
- snd_audigyls_ptr_write(chip, PLAYBACK_VOLUME, 0, 0x30303030);
-
- snd_audigyls_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
+ //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, 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, 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 */
+ snd_audigyls_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
+ snd_audigyls_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
+ snd_audigyls_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
+ snd_audigyls_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
+ 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, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
+
outl(0, chip->port+0x18);
snd_audigyls_intr_enable(chip, 0x105);
@@ -1276,10 +1352,14 @@
static int snd_audigyls_shared_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ static char *texts[2] = { "Digital", "Analog " };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 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]);
return 0;
}
@@ -1288,24 +1368,34 @@
{
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
- ucontrol->value.integer.value[0] = snd_audigyls_ptr_read(emu, SPDIF_SELECT, 0);
- return 0;
+ ucontrol->value.enumerated.item[0] = emu->digital_analog;
+ return 0;
}
static int snd_audigyls_shared_spdif_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
- unsigned int reg, val;
+ unsigned int val;
int change = 0;
- reg = snd_audigyls_ptr_read(emu, SPDIF_SELECT, 0);
- val = ucontrol->value.integer.value[0] ;
- /* FIXME: Check whether we want ON == SPDIF on, or OFF == SPDIF on */
- change = ((reg == 0xf00) != val);
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->digital_analog != val);
if (change) {
- reg ^= 0xf00;
- snd_audigyls_ptr_write(emu, SPDIF_SELECT, 0, reg);
+ emu->digital_analog = val;
+ if (val == 0) {
+ /* 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);
+ } else {
+ /* Analog */
+ snd_audigyls_ptr_write(emu, SPDIF_SELECT1, 0, 0xf00);
+ 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);
+ }
}
return change;
}
@@ -1417,16 +1507,19 @@
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 channel_id)
+ snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
{
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int value;
- value = snd_audigyls_ptr_read(emu, PLAYBACK_VOLUME, channel_id);
+ value = snd_audigyls_ptr_read(emu, reg, channel_id);
ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
return 0;
@@ -1436,68 +1529,91 @@
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_FRONT_CHANNEL;
- return snd_audigyls_volume_get(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_CENTER_LFE_CHANNEL;
- return snd_audigyls_volume_get(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_UNKNOWN_CHANNEL;
- return snd_audigyls_volume_get(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_REAR_CHANNEL;
- return snd_audigyls_volume_get(kcontrol, ucontrol, channel_id);
+ int reg = PLAYBACK_VOLUME1;
+ 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)
+{
+ int channel_id = 1;
+ int reg = CAPTURE_CONTROL;
+ return snd_audigyls_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_audigyls_volume_put(snd_kcontrol_t * kcontrol,
- snd_ctl_elem_value_t * ucontrol, int channel_id)
+ snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
{
audigyls_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int value;
- value = snd_audigyls_ptr_read(emu, PLAYBACK_VOLUME, channel_id);
- value = value & 0xffff;
- value = value | ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16);
- snd_audigyls_ptr_write(emu, PLAYBACK_VOLUME, channel_id, value);
+ //value = snd_audigyls_ptr_read(emu, reg, channel_id);
+ //value = value & 0xffff;
+ value = ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16);
+ value = value | ((0xff - ucontrol->value.integer.value[0]) << 8) | ((0xff - ucontrol->value.integer.value[1]) );
+ snd_audigyls_ptr_write(emu, reg, channel_id, value);
return 1;
}
static int snd_audigyls_volume_put_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_FRONT_CHANNEL;
- return snd_audigyls_volume_put(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_CENTER_LFE_CHANNEL;
- return snd_audigyls_volume_put(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_UNKNOWN_CHANNEL;
- return snd_audigyls_volume_put(kcontrol, ucontrol, channel_id);
+ 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,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = PCM_REAR_CHANNEL;
- return snd_audigyls_volume_put(kcontrol, ucontrol, channel_id);
+ int reg = PLAYBACK_VOLUME1;
+ 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)
+{
+ int channel_id = 1;
+ int reg = CAPTURE_CONTROL;
+ return snd_audigyls_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static snd_kcontrol_new_t snd_audigyls_volume_control_front =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Front Volume",
+ .name = "PCM Front Volume",
.info = snd_audigyls_volume_info_front,
.get = snd_audigyls_volume_get_front,
.put = snd_audigyls_volume_put_front
@@ -1505,7 +1621,7 @@
static snd_kcontrol_new_t snd_audigyls_volume_control_center_lfe =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Center/LFE Volume",
+ .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
@@ -1513,7 +1629,7 @@
static snd_kcontrol_new_t snd_audigyls_volume_control_unknown =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Unknown Volume",
+ .name = "PCM Unknown Volume",
.info = snd_audigyls_volume_info_unknown,
.get = snd_audigyls_volume_get_unknown,
.put = snd_audigyls_volume_put_unknown
@@ -1521,11 +1637,20 @@
static snd_kcontrol_new_t snd_audigyls_volume_control_rear =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Rear Volume",
+ .name = "PCM Rear Volume",
.info = snd_audigyls_volume_info_rear,
.get = snd_audigyls_volume_get_rear,
.put = snd_audigyls_volume_put_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,
+ .get = snd_audigyls_volume_get_feedback,
+ .put = snd_audigyls_volume_put_feedback
+};
+
static int remove_ctl(snd_card_t *card, const char *name)
{
@@ -1579,6 +1704,10 @@
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
+ if ((kctl = snd_ctl_new1(&snd_audigyls_volume_control_feedback, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
if ((kctl = snd_ctl_new1(&snd_audigyls_spdif_mask_control, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
@@ -1645,13 +1774,11 @@
snd_card_free(card);
return err;
}
-#if 1
if ((err = snd_audigyls_mixer(chip)) < 0) {
snd_card_free(card);
return err;
}
-#endif
snd_audigyls_proc_init(chip);
[-- Attachment #3: alsa-lib.audigyls.diff.txt --]
[-- Type: text/plain, Size: 1139 bytes --]
Index: alsa-lib/src/conf/cards/AudigyLS.conf
===================================================================
RCS file: /cvsroot/alsa/alsa-lib/src/conf/cards/AudigyLS.conf,v
retrieving revision 1.2
diff -u -r1.2 AudigyLS.conf
--- alsa-lib/src/conf/cards/AudigyLS.conf 23 Jun 2004 18:09:10 -0000 1.2
+++ alsa-lib/src/conf/cards/AudigyLS.conf 25 Jun 2004 20:18:10 -0000
@@ -23,7 +23,7 @@
}
type hw
card $CARD
- device 3
+ device 1
}
<confdir:pcm/center_lfe.conf>
@@ -35,7 +35,7 @@
}
type hw
card $CARD
- device 1
+ device 2
}
<confdir:pcm/surround40.conf>
@@ -146,12 +146,19 @@
slave.pcm {
type hw
card $CARD
- device 3
+ device 0
}
hooks.0 {
type ctl_elems
hook_args [
{
+ name "PCM Front Volume"
+ index 0
+ lock true
+ preserve true
+ value [ 207 207 ] # Puts 0x30303030 in the Volume register. 0xff - 0x30 = 0xcf = 207
+ }
+ {
name "Analog/Digital Output Jack"
lock true
preserve true
@@ -159,7 +166,7 @@
}
{
name "IEC958 Playback Default"
- index 0
+ index 1
lock true
preserve true
value [ $AES0 $AES1 $AES2 $AES3 ]
^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [PATCH] AudigyLS updates.
2004-06-25 21:37 [PATCH] AudigyLS updates James Courtier-Dutton
@ 2004-06-28 10:48 ` Takashi Iwai
2004-06-28 12:39 ` James Courtier-Dutton
0 siblings, 1 reply; 6+ messages in thread
From: Takashi Iwai @ 2004-06-28 10:48 UTC (permalink / raw)
To: James Courtier-Dutton; +Cc: ALSA development
At Fri, 25 Jun 2004 22:37:54 +0100,
James Courtier-Dutton wrote:
>
> AudigyLS now supports Digital(AC3)/Analog outputs. 5.1 Surround
> PLAYBACK. MIC and LINE IN CAPTURE.
>
> The MIC and LINE IN use the AC97 chip.
>
> The AC97 uses the PLAYBACK(Not CAPTURE) controls in alsamixer for
> controlling CAPTURE. Then, instead of sending that output to the
> speakers, it sends it to the Philips 1361T ADC. This results in high
> quality ADC conversion.
> E.g. For LINE IN capture, set LINE IN playback volume and unmute it.
> Set MASTER volume.
> There is a new control for "CAPTURE feedback into PLAYBACK". If one does
> not want any feedback, leave it at it's minimum.
>
> Once I have figured out all the controls in the AC97 and what they do
> for the AudigyLS, I will rename them so that they make sense to the user.
>
> Please see attached patch files against the current CVS.
Thanks, applied to cvs now.
> 2) Add a CAPUTRE source switch for switch between "MIC", "Line in",
> "SPDIF in", "TAD in", "AUX in" and "What u hear". Currently, one cannot
> switch to capture "SPDIF in" or "What u hear", but all the other inputs
> are possible via switches in the AC97.
This will need better control on user-space.
ac97 mixer will provide the generic capture source only...
Takashi
-------------------------------------------------------
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^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [PATCH] AudigyLS updates.
2004-06-28 10:48 ` Takashi Iwai
@ 2004-06-28 12:39 ` James Courtier-Dutton
2004-06-30 16:39 ` Takashi Iwai
0 siblings, 1 reply; 6+ messages in thread
From: James Courtier-Dutton @ 2004-06-28 12:39 UTC (permalink / raw)
To: Takashi Iwai; +Cc: ALSA development
Takashi Iwai wrote:
>
>>2) Add a CAPUTRE source switch for switch between "MIC", "Line in",
>>"SPDIF in", "TAD in", "AUX in" and "What u hear". Currently, one cannot
>>switch to capture "SPDIF in" or "What u hear", but all the other inputs
>>are possible via switches in the AC97.
>
>
> This will need better control on user-space.
> ac97 mixer will provide the generic capture source only...
>
>
> Takashi
>
>
I am adding better controls as I find out what does what.
How is the userspace mixer controls status? (Using a lisp .conf file to
control how the mixer behaves).
What shall I call the AC97 capture controls?
1) AC97 Master CAPTURE.
2) AC97 MIC CAPTURE.
3) AC97 Line in CAPTURE.
4) AC97 TAD in CAPTURE.
5) AC97 AUX CAPTURE.
option (1) to (5) can act together, so one can record some from MIC and
also some from Line in at the same time.
6) Feedback into PLAYBACK from CAPTURE.
There is also another capture source register I have found out about.
0 - Selects SPDIF CAPTURE.
1 - What you hear CAPTURE.
2 - Unknown
3 - AC97 CAPTURE. (See 1) to 5) above)
If this register is not set to 3, all the AC97 inputs are muted.
I think I will add some new control renaming/deleting code, so that only
the controls I know about are displayed to the user. I can put an #if 1
... #endif round it, so that I can disable the renaming when I am testing.
Is there a document somewhere for alsa-driver developers that states
what one should set things to as the default, the "on start up" settings?
E.g.
Initial Analog volume controls.
Should digital out be enabled at startup, or analog outputs.
I think we should have the following (although the audigyls.c does not
follow this yet!):-
Analog volume controls muted.
Default to Analog output.
In windows the defaults are:
I don't know what the analog defaults are.
digital out is disabled at start up.
The windows driver controls digital output using a on/off switch called
"SPDIF Out"
I think it might be a good idea to change the "Audigy Analog/Digital
Output Jack" to instead say "SPDIF Out", with it defaulting to off for
Analog.
The "Audigy Analog/Digital Output Jack" being an on/off switch was
confusing, because it was not clear to a casual user what [off] meant.
Does that mean "Analog off", or "Digital off".
Cheers
James
-------------------------------------------------------
This SF.Net email sponsored by Black Hat Briefings & Training.
Attend Black Hat Briefings & Training, Las Vegas July 24-29 -
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^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [PATCH] AudigyLS updates.
2004-06-28 12:39 ` James Courtier-Dutton
@ 2004-06-30 16:39 ` Takashi Iwai
2004-07-01 12:34 ` James Courtier-Dutton
0 siblings, 1 reply; 6+ messages in thread
From: Takashi Iwai @ 2004-06-30 16:39 UTC (permalink / raw)
To: James Courtier-Dutton; +Cc: ALSA development
At Mon, 28 Jun 2004 13:39:25 +0100,
James Courtier-Dutton wrote:
>
> Takashi Iwai wrote:
> >
> >>2) Add a CAPUTRE source switch for switch between "MIC", "Line in",
> >>"SPDIF in", "TAD in", "AUX in" and "What u hear". Currently, one cannot
> >>switch to capture "SPDIF in" or "What u hear", but all the other inputs
> >>are possible via switches in the AC97.
> >
> >
> > This will need better control on user-space.
> > ac97 mixer will provide the generic capture source only...
> >
> >
> > Takashi
> >
> >
>
> I am adding better controls as I find out what does what.
> How is the userspace mixer controls status? (Using a lisp .conf file to
> control how the mixer behaves).
Jaroslav has been working on this, but I think it's not ready yet...
> What shall I call the AC97 capture controls?
> 1) AC97 Master CAPTURE.
> 2) AC97 MIC CAPTURE.
> 3) AC97 Line in CAPTURE.
> 4) AC97 TAD in CAPTURE.
> 5) AC97 AUX CAPTURE.
> option (1) to (5) can act together, so one can record some from MIC and
> also some from Line in at the same time.
In the case of SB16, it's simply like 'Mic Capture Switch'.
> 6) Feedback into PLAYBACK from CAPTURE.
Isn't it 'what u hear'? What's different from 'master capture'?
> There is also another capture source register I have found out about.
> 0 - Selects SPDIF CAPTURE.
> 1 - What you hear CAPTURE.
> 2 - Unknown
> 3 - AC97 CAPTURE. (See 1) to 5) above)
> If this register is not set to 3, all the AC97 inputs are muted.
>
> I think I will add some new control renaming/deleting code, so that only
> the controls I know about are displayed to the user. I can put an #if 1
> ... #endif round it, so that I can disable the renaming when I am testing.
The simplest way would be to implement enum as the capture source as
an exclusive capture source. Of course the only drawback is that one
cannot choose Line and CD at the same time in this case, but I don't
think not many people will complain against this restriction.
> Is there a document somewhere for alsa-driver developers that states
> what one should set things to as the default, the "on start up" settings?
No, it's up to the device.
In general, we choose the mute as default, but I think not all volumes
have to be muted (e.g. only PCM and Master).
> E.g.
> Initial Analog volume controls.
> Should digital out be enabled at startup, or analog outputs.
>
> I think we should have the following (although the audigyls.c does not
> follow this yet!):-
> Analog volume controls muted.
> Default to Analog output.
>
> In windows the defaults are:
> I don't know what the analog defaults are.
> digital out is disabled at start up.
IMO, these initial state can be handled better from the user-space,
once when we have dB-based volume controls.
> The windows driver controls digital output using a on/off switch called
> "SPDIF Out"
>
> I think it might be a good idea to change the "Audigy Analog/Digital
> Output Jack" to instead say "SPDIF Out", with it defaulting to off for
> Analog.
Well, this is a switch for the shared output jack, so "SPDIF Out"
would be too ambiguous for its purpose. But I agree with the renaming
to a more reasonable one.
> The "Audigy Analog/Digital Output Jack" being an on/off switch was
> confusing, because it was not clear to a casual user what [off] meant.
> Does that mean "Analog off", or "Digital off".
Yes.
Takashi
-------------------------------------------------------
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,
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^ permalink raw reply [flat|nested] 6+ messages in thread
* Re: [PATCH] AudigyLS updates.
2004-06-30 16:39 ` Takashi Iwai
@ 2004-07-01 12:34 ` James Courtier-Dutton
2004-07-01 13:30 ` Takashi Iwai
0 siblings, 1 reply; 6+ messages in thread
From: James Courtier-Dutton @ 2004-07-01 12:34 UTC (permalink / raw)
To: Takashi Iwai; +Cc: ALSA development
[-- Attachment #1: Type: text/plain, Size: 5165 bytes --]
Takashi Iwai wrote:
> At Mon, 28 Jun 2004 13:39:25 +0100,
> James Courtier-Dutton wrote:
>
>>Takashi Iwai wrote:
>>
>>>>2) Add a CAPUTRE source switch for switch between "MIC", "Line in",
>>>>"SPDIF in", "TAD in", "AUX in" and "What u hear". Currently, one cannot
>>>>switch to capture "SPDIF in" or "What u hear", but all the other inputs
>>>>are possible via switches in the AC97.
>>>
>>>
>>>This will need better control on user-space.
>>>ac97 mixer will provide the generic capture source only...
>>>
>>>
>>>Takashi
>>>
>>>
>>
>>I am adding better controls as I find out what does what.
>>How is the userspace mixer controls status? (Using a lisp .conf file to
>>control how the mixer behaves).
>
>
> Jaroslav has been working on this, but I think it's not ready yet...
>
>
>>What shall I call the AC97 capture controls?
>>1) AC97 Master CAPTURE.
>>2) AC97 MIC CAPTURE.
>>3) AC97 Line in CAPTURE.
>>4) AC97 TAD in CAPTURE.
>>5) AC97 AUX CAPTURE.
>>option (1) to (5) can act together, so one can record some from MIC and
>>also some from Line in at the same time.
>
>
> In the case of SB16, it's simply like 'Mic Capture Switch'.
>
>
>>6) Feedback into PLAYBACK from CAPTURE.
>
>
> Isn't it 'what u hear'? What's different from 'master capture'?
master capture is the renamed AC97 Master playback, but with the
audigyls, it only effects the recorded signal. So, for recording, we
have separate Line in, Mic volume controls, and then a master one
covering all of them. As one is only likely to want to record from one
source at a time, I could just remove the "AC97 Master capture" control,
and set it at 0dB.
"Feedback into PLAYBACK from CAPTURE" is a way to send any recorded
sounds, e.g. MIC, directly to the speakers, without them reaching the CPU.
"what u hear" is a way to capture the sound exactly as it appears on the
speakers. i.e. the final mix of what your are currently playing combined
with any thing from "feedback into playback from capture".
>
>
>>There is also another capture source register I have found out about.
>>0 - Selects SPDIF CAPTURE.
>>1 - What you hear CAPTURE.
>>2 - Unknown
>>3 - AC97 CAPTURE. (See 1) to 5) above)
>>If this register is not set to 3, all the AC97 inputs are muted.
>>
>>I think I will add some new control renaming/deleting code, so that only
>>the controls I know about are displayed to the user. I can put an #if 1
>>... #endif round it, so that I can disable the renaming when I am testing.
>
>
> The simplest way would be to implement enum as the capture source as
> an exclusive capture source. Of course the only drawback is that one
> cannot choose Line and CD at the same time in this case, but I don't
> think not many people will complain against this restriction.
>
>
>
>>Is there a document somewhere for alsa-driver developers that states
>>what one should set things to as the default, the "on start up" settings?
>
>
> No, it's up to the device.
> In general, we choose the mute as default, but I think not all volumes
> have to be muted (e.g. only PCM and Master).
>
>
>>E.g.
>>Initial Analog volume controls.
>>Should digital out be enabled at startup, or analog outputs.
>>
>>I think we should have the following (although the audigyls.c does not
>>follow this yet!):-
>>Analog volume controls muted.
>>Default to Analog output.
>>
>>In windows the defaults are:
>>I don't know what the analog defaults are.
>>digital out is disabled at start up.
>
>
> IMO, these initial state can be handled better from the user-space,
> once when we have dB-based volume controls.
>
>
>>The windows driver controls digital output using a on/off switch called
>>"SPDIF Out"
>>
>>I think it might be a good idea to change the "Audigy Analog/Digital
>>Output Jack" to instead say "SPDIF Out", with it defaulting to off for
>>Analog.
>
>
> Well, this is a switch for the shared output jack, so "SPDIF Out"
> would be too ambiguous for its purpose. But I agree with the renaming
> to a more reasonable one.
I think "SPDIF out" is not too ambiguous.
If "SPDIF Out" is off, you are not doing digital output, so one can
switch the shared jack to analog.
If "SPDIF Out" in on, you want to use digital output, so one can switch
the shared jack to digital.
I think that a even better approach would be using a
"Speaker-arrangement" control.
Options:
"2.0"
"4.0"
"4.1"
"5.0"
"5.1"
...
"SPDIF digital output."
So, if someone set it to "2.0", it would remove any mixer controls that
controlled the rear and center_lfe channels.
I expect that will be best done in user space with the lisp .conf files
when they are ready.
>
>
>>The "Audigy Analog/Digital Output Jack" being an on/off switch was
>>confusing, because it was not clear to a casual user what [off] meant.
>>Does that mean "Analog off", or "Digital off".
>
>
> Yes.
>
>
> Takashi
>
>
I attach a new patch for the current CVS.
My CVS is delayed with respect to yours, so your latest MODULE...
changes are not included, so the patch might fail.
I still need to implement your suggestion for moving the periods list to
card DMA, instead of using the substream DMA buffer.
Cheers
James
[-- Attachment #2: audigyls1.diff.txt --]
[-- Type: text/plain, Size: 35323 bytes --]
Index: alsa-driver/pci/emu10k1/audigyls.c
===================================================================
RCS file: /cvsroot/alsa/alsa-driver/pci/emu10k1/audigyls.c,v
retrieving revision 1.4
diff -u -r1.4 audigyls.c
--- alsa-driver/pci/emu10k1/audigyls.c 29 Jun 2004 16:16:29 -0000 1.4
+++ alsa-driver/pci/emu10k1/audigyls.c 1 Jul 2004 12:14:16 -0000
@@ -1,6 +1,7 @@
/*
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver AUDIGYLS chips
+ * Version: 0.5
*
* FEATURES currently supported:
* Front, Rear and Center/LFE.
@@ -17,16 +18,37 @@
* 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.
+ *
*
* BUGS:
* Some stability problems when unloading the snd-audigyls kernel module.
* --
*
* TODO:
- * Need to add a way to select capture source.
+ * Playback period list using card based DMA instead of top bit of substream DMA.
* 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
* --
@@ -111,6 +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 */
@@ -118,19 +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 ?? */
@@ -138,10 +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) */
@@ -193,6 +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 */
@@ -246,6 +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;
@@ -301,7 +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;
@@ -320,8 +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),
@@ -644,18 +684,26 @@
{
audigyls_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
+ u32 *table_base = (u32 *)(runtime->dma_area+(32*1024));
audigyls_pcm_t *epcm = runtime->private_data;
- int voice = epcm->voice->number;
- voice=epcm->channel_id;
+ int voice = voice=epcm->channel_id;
+ 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);
+ 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, runtime->dma_addr + (32*1024));
+ 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
@@ -756,15 +804,19 @@
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;
@@ -905,7 +957,8 @@
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
@@ -957,7 +1010,7 @@
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++) {
@@ -981,7 +1034,7 @@
}
}
- 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);
@@ -1041,7 +1094,7 @@
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;
}
@@ -1192,12 +1245,13 @@
//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 */
@@ -1208,16 +1262,20 @@
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);
@@ -1349,14 +1407,10 @@
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;
}
@@ -1365,7 +1419,7 @@
{
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;
}
@@ -1375,23 +1429,29 @@
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;
@@ -1400,12 +1460,63 @@
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;
@@ -1476,7 +1587,7 @@
.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;
@@ -1484,31 +1595,6 @@
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)
@@ -1522,35 +1608,65 @@
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)
{
@@ -1571,34 +1687,63 @@
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)
{
@@ -1607,43 +1752,76 @@
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 = "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_analog_unknown =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .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_analog_rear =
{
.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 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_center_lfe =
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_front =
{
.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 = "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_unknown =
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_center_lfe =
{
.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 = "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_rear =
+static snd_kcontrol_new_t snd_audigyls_volume_control_spdif_unknown =
{
.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 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
};
@@ -1668,7 +1846,6 @@
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);
@@ -1678,26 +1855,76 @@
}
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)
+#if 1
+ remove_ctl(card, "Master Mono Playback Switch");
+ remove_ctl(card, "Master Mono Playback Volume");
+ remove_ctl(card, "3D Control - Switch");
+ remove_ctl(card, "3D Control Sigmatel - Depth");
+ remove_ctl(card, "PCM Playback Switch");
+ remove_ctl(card, "PCM Playback Volume");
+ remove_ctl(card, "CD Playback Switch");
+ remove_ctl(card, "CD Playback Volume");
+ remove_ctl(card, "Phone Playback Switch");
+ remove_ctl(card, "Phone Playback Volume");
+ remove_ctl(card, "Video Playback Switch");
+ remove_ctl(card, "Video Playback Volume");
+ remove_ctl(card, "PC Speaker Playback Switch");
+ remove_ctl(card, "PC Speaker Playback Volume");
+ remove_ctl(card, "Mono Output Select");
+ remove_ctl(card, "Capture Source");
+ remove_ctl(card, "Capture Switch");
+ remove_ctl(card, "Capture Volume");
+ remove_ctl(card, "External Amplifier");
+ remove_ctl(card, "Sigmatel 4-Speaker Stereo Playback Switch");
+ remove_ctl(card, "Sigmatel Surround Phase Inversion Playback ");
+ rename_ctl(card, "Master Playback Switch", "Capture Switch");
+ rename_ctl(card, "Master Playback Volume", "Capture Volume");
+ rename_ctl(card, "Line Playback Switch", "AC97 Line Capture Switch");
+ rename_ctl(card, "Line Playback Volume", "AC97 Line Capture Volume");
+ rename_ctl(card, "Aux Playback Switch", "AC97 Aux Capture Switch");
+ rename_ctl(card, "Aux Playback Volume", "AC97 Aux Capture Volume");
+ rename_ctl(card, "Mic Playback Switch", "AC97 Mic Capture Switch");
+ rename_ctl(card, "Mic Playback Volume", "AC97 Mic Capture Volume");
+ rename_ctl(card, "Mic Select", "AC97 Mic Select");
+ rename_ctl(card, "Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)");
+ //rename_ctl(card, "", "");
+#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;
@@ -1713,6 +1940,10 @@
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] 6+ messages in thread
* Re: [PATCH] AudigyLS updates.
2004-07-01 12:34 ` James Courtier-Dutton
@ 2004-07-01 13:30 ` Takashi Iwai
0 siblings, 0 replies; 6+ messages in thread
From: Takashi Iwai @ 2004-07-01 13:30 UTC (permalink / raw)
To: James Courtier-Dutton; +Cc: ALSA development
At Thu, 01 Jul 2004 13:34:17 +0100,
James Courtier-Dutton wrote:
>
> >>6) Feedback into PLAYBACK from CAPTURE.
> >
> >
> > Isn't it 'what u hear'? What's different from 'master capture'?
>
> master capture is the renamed AC97 Master playback, but with the
> audigyls, it only effects the recorded signal. So, for recording, we
> have separate Line in, Mic volume controls, and then a master one
> covering all of them. As one is only likely to want to record from one
> source at a time, I could just remove the "AC97 Master capture" control,
> and set it at 0dB.
>
> "Feedback into PLAYBACK from CAPTURE" is a way to send any recorded
> sounds, e.g. MIC, directly to the speakers, without them reaching the CPU.
>
> "what u hear" is a way to capture the sound exactly as it appears on the
> speakers. i.e. the final mix of what your are currently playing combined
> with any thing from "feedback into playback from capture".
Well then 'what u hear' suffices most of user's purpose.
> I think that a even better approach would be using a
> "Speaker-arrangement" control.
> Options:
> "2.0"
> "4.0"
> "4.1"
> "5.0"
> "5.1"
> ...
> "SPDIF digital output."
>
> So, if someone set it to "2.0", it would remove any mixer controls that
> controlled the rear and center_lfe channels.
> I expect that will be best done in user space with the lisp .conf files
It's a good idea.
> I attach a new patch for the current CVS.
> My CVS is delayed with respect to yours, so your latest MODULE...
> changes are not included, so the patch might fail.
It's no problem, patch is ok unless it includes 'MODULE_*' in it.
> + remove_ctl(card, "Master Mono Playback Switch");
> + remove_ctl(card, "Master Mono Playback Volume");
> + remove_ctl(card, "3D Control - Switch");
> + remove_ctl(card, "3D Control Sigmatel - Depth");
(snip)
Could you make an array and call remove_ctl() in a loop instead of
each one?
Takashi
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^ permalink raw reply [flat|nested] 6+ messages in thread
end of thread, other threads:[~2004-07-01 13:32 UTC | newest]
Thread overview: 6+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2004-06-25 21:37 [PATCH] AudigyLS updates James Courtier-Dutton
2004-06-28 10:48 ` Takashi Iwai
2004-06-28 12:39 ` James Courtier-Dutton
2004-06-30 16:39 ` Takashi Iwai
2004-07-01 12:34 ` James Courtier-Dutton
2004-07-01 13:30 ` Takashi Iwai
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