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-rw-r--r--sound/pci/emu10k1/emu10k1_main.c52
-rw-r--r--sound/pci/emu10k1/emufx.c78
-rw-r--r--sound/pci/emu10k1/emumixer.c16
-rw-r--r--sound/pci/emu10k1/emupcm.c39
4 files changed, 172 insertions, 13 deletions
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 4a9b59ad8ab..72367411ee6 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -694,6 +694,37 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file
return 0;
}
+/*
+ * EMU-1010 - details found out from this driver, official MS Win drivers,
+ * testing the card:
+ *
+ * Audigy2 (aka Alice2):
+ * ---------------------
+ * * communication over PCI
+ * * conversion of 32-bit data coming over EMU32 links from HANA FPGA
+ * to 2 x 16-bit, using internal DSP instructions
+ * * slave mode, clock supplied by HANA
+ * * linked to HANA using:
+ * 32 x 32-bit serial EMU32 output channels
+ * 16 x EMU32 input channels
+ * (?) x I2S I/O channels (?)
+ *
+ * FPGA (aka HANA):
+ * ---------------
+ * * provides all (?) physical inputs and outputs of the card
+ * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.)
+ * * provides clock signal for the card and Alice2
+ * * two crystals - for 44.1kHz and 48kHz multiples
+ * * provides internal routing of signal sources to signal destinations
+ * * inputs/outputs to Alice2 - see above
+ *
+ * Current status of the driver:
+ * ----------------------------
+ * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz)
+ * * PCM device nb. 2:
+ * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops
+ * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops
+ */
static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
{
unsigned int i;
@@ -850,6 +881,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu)
EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1);
snd_emu1010_fpga_link_dst_src_write(emu,
EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1);
+ /* Pavel Hofman - setting defaults for 8 more capture channels
+ * Defaults only, users will set their own values anyways, let's
+ * just copy/paste.
+ */
+
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1);
+ snd_emu1010_fpga_link_dst_src_write(emu,
+ EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1);
#endif
#if 0
/* Original */
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index c02012cccd8..7206c0fa06f 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1123,6 +1123,11 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl
ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF;
}
+/*
+ * Used for emu1010 - conversion from 32-bit capture inputs from HANA
+ * to 2 x 16-bit registers in audigy - their values are read via DMA.
+ * Conversion is performed by Audigy DSP instructions of FX8010.
+ */
static int snd_emu10k1_audigy_dsp_convert_32_to_2x16(
struct snd_emu10k1_fx8010_code *icode,
u32 *ptr, int tmp, int bit_shifter16,
@@ -1193,7 +1198,11 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu)
snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP);
#if 1
- /* PCM front Playback Volume (independent from stereo mix) */
+ /* PCM front Playback Volume (independent from stereo mix)
+ * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31)
+ * where gpr contains attenuation from corresponding mixer control
+ * (snd_emu10k1_init_stereo_control)
+ */
A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT));
A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT));
snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100);
@@ -1549,7 +1558,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
if (emu->card_capabilities->emu1010) {
snd_printk("EMU inputs on\n");
- /* Capture 8 channels of S32_LE sound */
+ /* Capture 16 (originally 8) channels of S32_LE sound */
/* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
/* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
@@ -1560,6 +1569,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) );
/* Right ADC in 1 of 2 */
gpr_map[gpr++] = 0x00000000;
+ /* Delaying by one sample: instead of copying the input
+ * value A_P16VIN to output A_FXBUS2 as in the first channel,
+ * we use an auxiliary register, delaying the value by one
+ * sample
+ */
snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) );
A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000);
gpr_map[gpr++] = 0x00000000;
@@ -1583,6 +1597,66 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
gpr_map[gpr++] = 0x00000000;
snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) );
A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000);
+ /* Pavel Hofman - we still have voices, A_FXBUS2s, and
+ * A_P16VINs available -
+ * let's add 8 more capture channels - total of 16
+ */
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x10));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x12));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x14));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x16));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x18));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1a));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1c));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe),
+ A_C_00000000, A_C_00000000);
+ gpr_map[gpr++] = 0x00000000;
+ snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp,
+ bit_shifter16,
+ A_GPR(gpr - 1),
+ A_FXBUS2(0x1e));
+ A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf),
+ A_C_00000000, A_C_00000000);
#if 0
for (z = 4; z < 8; z++) {
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 4db6e1ca166..7b2c1dcc533 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -77,6 +77,10 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol,
return 0;
}
+/*
+ * Items labels in enum mixer controls assigning source data to
+ * each destination
+ */
static char *emu1010_src_texts[] = {
"Silence",
"Dock Mic A",
@@ -133,6 +137,9 @@ static char *emu1010_src_texts[] = {
"DSP 31",
};
+/*
+ * List of data sources available for each destination
+ */
static unsigned int emu1010_src_regs[] = {
EMU_SRC_SILENCE,/* 0 */
EMU_SRC_DOCK_MIC_A1, /* 1 */
@@ -189,6 +196,10 @@ static unsigned int emu1010_src_regs[] = {
EMU_SRC_ALICE_EMU32B+0xf, /* 52 */
};
+/*
+ * Data destinations - physical EMU outputs.
+ * Each destination has an enum mixer control to choose a data source
+ */
static unsigned int emu1010_output_dst[] = {
EMU_DST_DOCK_DAC1_LEFT1, /* 0 */
EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */
@@ -216,6 +227,11 @@ static unsigned int emu1010_output_dst[] = {
EMU_DST_HANA_ADAT+7, /* 23 */
};
+/*
+ * Data destinations - HANA outputs going to Alice2 (audigy) for
+ * capture (EMU32 + I2S links)
+ * Each destination has an enum mixer control to choose a data source
+ */
static unsigned int emu1010_input_dst[] = {
EMU_DST_ALICE2_EMU32_0,
EMU_DST_ALICE2_EMU32_1,
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index ab4f5df5241..eda5cb373de 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1233,24 +1233,26 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
runtime->hw.rate_min = runtime->hw.rate_max = 48000;
spin_lock_irq(&emu->reg_lock);
if (emu->card_capabilities->emu1010) {
- /* TODO
+ /* Nb. of channels has been increased to 16 */
+ /* TODO
* SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE
* SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
* SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
* SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000
* rate_min = 44100,
* rate_max = 192000,
- * channels_min = 8,
- * channels_max = 8,
+ * channels_min = 16,
+ * channels_max = 16,
* Need to add mixer control to fix sample rate
*
- * There are 16 mono channels of 16bits each.
+ * There are 32 mono channels of 16bits each.
* 24bit Audio uses 2x channels over 16bit
* 96kHz uses 2x channels over 48kHz
* 192kHz uses 4x channels over 48kHz
- * So, for 48kHz 24bit, one has 8 channels
- * for 96kHz 24bit, one has 4 channels
- * for 192kHz 24bit, one has 2 channels
+ * So, for 48kHz 24bit, one has 16 channels
+ * for 96kHz 24bit, one has 8 channels
+ * for 192kHz 24bit, one has 4 channels
+ *
*/
#if 1
switch (emu->emu1010.internal_clock) {
@@ -1258,13 +1260,15 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
/* For 44.1kHz */
runtime->hw.rates = SNDRV_PCM_RATE_44100;
runtime->hw.rate_min = runtime->hw.rate_max = 44100;
- runtime->hw.channels_min = runtime->hw.channels_max = 8;
+ runtime->hw.channels_min =
+ runtime->hw.channels_max = 16;
break;
case 1:
/* For 48kHz */
runtime->hw.rates = SNDRV_PCM_RATE_48000;
runtime->hw.rate_min = runtime->hw.rate_max = 48000;
- runtime->hw.channels_min = runtime->hw.channels_max = 8;
+ runtime->hw.channels_min =
+ runtime->hw.channels_max = 16;
break;
};
#endif
@@ -1282,7 +1286,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream)
#endif
runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
/* efx_voices_mask[0] is expected to be zero
- * efx_voices_mask[1] is expected to have 16bits set
+ * efx_voices_mask[1] is expected to have 32bits set
*/
} else {
runtime->hw.channels_min = runtime->hw.channels_max = 0;
@@ -1787,11 +1791,24 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s
/* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */
if (emu->audigy) {
emu->efx_voices_mask[0] = 0;
- emu->efx_voices_mask[1] = 0xffff;
+ if (emu->card_capabilities->emu1010)
+ /* Pavel Hofman - 32 voices will be used for
+ * capture (write mode) -
+ * each bit = corresponding voice
+ */
+ emu->efx_voices_mask[1] = 0xffffffff;
+ else
+ emu->efx_voices_mask[1] = 0xffff;
} else {
emu->efx_voices_mask[0] = 0xffff0000;
emu->efx_voices_mask[1] = 0;
}
+ /* For emu1010, the control has to set 32 upper bits (voices)
+ * out of the 64 bits (voices) to true for the 16-channels capture
+ * to work correctly. Correct A_FXWC2 initial value (0xffffffff)
+ * is already defined but the snd_emu10k1_pcm_efx_voices_mask
+ * control can override this register's value.
+ */
kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu);
if (!kctl)
return -ENOMEM;