ALSA update

  - CS46xx
    - volume bug fixes and phase reversal fix
    - AC3 stuff...
    - SPDIF input fix
  - init.c
    - used workqueue for the disconnection job
  - CMIPCI
    - S/PDIF output / PLL updates
  - ICE1712
    - fixed reversed volume for AK4524 and AK4528 codecs
    - added support for Hoontech STA DSP24 Media 7.1
  - USB audio
    - various mixer fixes
    - quirks for Edirol PCR-30/50 keyboards and Midiman hardware

include/sound/cs46xx.h

@@ -1663,6 +1663,8 @@ typedef struct _snd_cs46xx_pcm_t {
 	snd_pcm_substream_t *substream;
 
 	pcm_channel_descriptor_t * pcm_channel;
+
+	int pcm_channel_id;    /* Fron Rear, Center Lfe  ... */
 } cs46xx_pcm_t;
 
 typedef struct {

include/sound/cs46xx_dsp_spos.h

@@ -140,7 +140,6 @@ typedef struct _pcm_channel_descriptor_t {
 	dsp_scb_descriptor_t * pcm_reader_scb;
 	dsp_scb_descriptor_t * src_scb;
 	dsp_scb_descriptor_t * mixer_scb;
-	int pcm_channel_id;
 
 	void * private_data;
 } pcm_channel_descriptor_t;

include/sound/version.h

 /* include/version.h.  Generated automatically by configure.  */
 #define CONFIG_SND_VERSION "0.9.0rc6"
-#define CONFIG_SND_DATE " (Thu Dec 05 10:04:08 2002 UTC)"
+#define CONFIG_SND_DATE " (Wed Dec 11 11:08:10 2002 UTC)"

sound/core/init.c

@@ -26,7 +26,7 @@
 #include <linux/slab.h>
 #include <linux/time.h>
 #include <linux/ctype.h>
-#include <linux/smp_lock.h>
+#include <linux/workqueue.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
@@ -152,13 +152,13 @@ int snd_card_disconnect(snd_card_t * card)
 	struct file_operations *f_ops, *old_f_ops;
 	int err;
 
-	write_lock(&card->files_lock);
+	spin_lock(&card->files_lock);
 	if (card->shutdown) {
-		write_unlock(&card->files_lock);
+		spin_unlock(&card->files_lock);
 		return 0;
 	}
 	card->shutdown = 1;
-	write_unlock(&card->files_lock);
+	spin_unlock(&card->files_lock);
 
 	/* phase 1: disable fops (user space) operations for ALSA API */
 	write_lock(&snd_card_rwlock);
@@ -181,7 +181,9 @@ int snd_card_disconnect(snd_card_t * card)
 		f_ops = &s_f_ops->f_ops;
 
 		memset(f_ops, 0, sizeof(*f_ops));
+#ifndef LINUX_2_2
 		f_ops->owner = file->f_op->owner;
+#endif
 		f_ops->release = file->f_op->release;
 		f_ops->poll = snd_disconnect_poll;
 
@@ -220,6 +222,10 @@ int snd_card_disconnect(snd_card_t * card)
  *  snd_card_free: frees given soundcard structure
  *  @card: soundcard structure
  *
+ *  This function releases the soundcard structure and the all assigned
+ *  devices automatically.  That is, you don't have to release the devices
+ *  by yourself.
+ *
  *  Returns - zero. Frees all associated devices and frees the control
  *  interface associated to given soundcard.
  */
@@ -283,22 +289,11 @@ int snd_card_free(snd_card_t * card)
 	return 0;
 }
 
-static int snd_card_free_thread(void * __card)
+static void snd_card_free_thread(void * __card)
 {
 	snd_card_t *card = __card;
 	struct module * module;
 
-#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0)
-	lock_kernel();
-#endif
-
-	daemonize();
-#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0)
-	reparent_to_init();
-#endif
-
-	strcpy(current->comm, "snd-free");
-
 	if (!try_inc_mod_count(module = card->module)) {
 		snd_printk(KERN_ERR "unable to lock toplevel module for card %i in free thread\n", card->number);
 		module = NULL;
@@ -310,31 +305,33 @@ static int snd_card_free_thread(void * __card)
 
 	if (module)
 		__MOD_DEC_USE_COUNT(module);
-
-#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 0)
-	unlock_kernel();
-#endif
-
-	return 0;
 }
 
 /**
  *  snd_card_free_in_thread: call snd_card_free() in thread
  *  @card: soundcard structure
  *
+ *  This function schedules the call of #snd_card_free function in a
+ *  work queue.  When all devices are released (non-busy), the work
+ *  is woken up and calls #snd_card_free.
+ *
+ *  When a card can be disconnected at any time by hotplug service,
+ *  this function should be used in disconnect (or detach) callback
+ *  instead of calling #snd_card_free directly.
+ *  
  *  Returns - zero otherwise a negative error code if the start of thread failed.
  */
 int snd_card_free_in_thread(snd_card_t * card)
 {
-	int pid;
+	DECLARE_WORK(works, snd_card_free_thread, card);
 
 	if (card->files == NULL) {
 		snd_card_free(card);
 		return 0;
 	}
-	pid = kernel_thread(snd_card_free_thread, card, 0);
-	if (pid >= 0)
+	if (schedule_work(&works))
 		return 0;
+
 	snd_printk(KERN_ERR "kernel_thread failed in snd_card_free_in_thread for card %i\n", card->number);
 	/* try to free the structure immediately */
 	snd_card_free(card);
@@ -395,6 +392,17 @@ static void choose_default_id(snd_card_t * card)
 	}
 }
 
+/**
+ *  snd_card_register: register the soundcard
+ *  @card: soundcard structure
+ *
+ *  This function registers all the devices assigned to the soundcard.
+ *  Until calling this, the ALSA control interface is blocked from the
+ *  external accesses.  Thus, you should call this function at the end
+ *  of the initialization of the card.
+ *
+ *  Returns - zero otherwise a negative error code if the registrain failed.
+ */
 int snd_card_register(snd_card_t * card)
 {
 	int err;
@@ -509,6 +517,17 @@ int __exit snd_card_info_done(void)
 	return 0;
 }
 
+/**
+ *  snd_component_add: add a component string
+ *  @card: soundcard structure
+ *  @component: the component id string
+ *
+ *  This function adds the component id string to the supported list.
+ *  The component can be referred from the alsa-lib.
+ *
+ *  Returns - zero otherwise a negative error code.
+ */
+  
 int snd_component_add(snd_card_t *card, const char *component)
 {
 	char *ptr;
@@ -529,6 +548,17 @@ int snd_component_add(snd_card_t *card, const char *component)
 	return 0;
 }
 
+/**
+ *  snd_card_file_add: add the file to the file list of the card
+ *  @card: soundcard structure
+ *  @file: file pointer
+ *
+ *  This function adds the file to the file linked-list of the card.
+ *  This linked-list is used to keep tracking the connection state,
+ *  and to avoid the release of busy resources by hotplug.
+ *
+ *  Returns zero or a negative error code.
+ */
 int snd_card_file_add(snd_card_t *card, struct file *file)
 {
 	struct snd_monitor_file *mfile;
@@ -550,6 +580,19 @@ int snd_card_file_add(snd_card_t *card, struct file *file)
 	return 0;
 }
 
+/**
+ *  snd_card_file_remove: remove the file from the file list
+ *  @card: soundcard structure
+ *  @file: file pointer
+ *
+ *  This function removes the file formerly added to the card via
+ *  #snd_card_file_add function.
+ *  If all files are removed and the release of the card is
+ *  scheduled, it will wake up the the thread to call #snd_card_free
+ *  (see #snd_card_free_in_thread function).
+ *
+ *  Returns zero or a negative error code.
+ */
 int snd_card_file_remove(snd_card_t *card, struct file *file)
 {
 	struct snd_monitor_file *mfile, *pfile = NULL;
@@ -580,7 +623,12 @@ int snd_card_file_remove(snd_card_t *card, struct file *file)
 }
 
 #ifdef CONFIG_PM
-/* the power lock must be active before call */
+/**
+ *  snd_power_wait: wait until the power-state is changed.
+ *  @card: soundcard structure
+ *
+ *  Note: the power lock must be active before call.
+ */
 void snd_power_wait(snd_card_t *card)
 {
 	wait_queue_t wait;

sound/core/pcm_lib.c

@@ -110,7 +110,7 @@ int snd_pcm_update_hw_ptr_interrupt(snd_pcm_substream_t *substream)
 		snd_printk(KERN_ERR  "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size);
 	} else
 #endif
-	snd_runtime_check(pos <= runtime->buffer_size, return 0);
+	snd_runtime_check(pos < runtime->buffer_size, return 0);
 
 	pos -= pos % runtime->min_align;
 	new_hw_ptr = runtime->hw_ptr_base + pos;
@@ -176,7 +176,7 @@ int snd_pcm_update_hw_ptr(snd_pcm_substream_t *substream)
 		snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size);
 	} else
 #endif
-	snd_runtime_check(pos <= runtime->buffer_size, return 0);
+	snd_runtime_check(pos < runtime->buffer_size, return 0);
 
 	pos -= pos % runtime->min_align;
 	new_hw_ptr = runtime->hw_ptr_base + pos;

sound/core/seq/seq_device.c

@@ -283,6 +283,8 @@ static int snd_seq_device_dev_disconnect(snd_device_t *device)
 		return -ENOENT;
 
 	free_device(dev, ops);
+
+	unlock_driver(ops);
 	return 0;
 }
 

sound/isa/cs423x/cs4231_lib.c

@@ -445,6 +445,9 @@ static int snd_cs4231_trigger(snd_pcm_substream_t *substream,
 {
 	cs4231_t *chip = snd_pcm_substream_chip(substream);
 	int result = 0;
+	unsigned int what;
+	snd_pcm_substream_t *s;
+	int do_start;
 
 #if 0
 	printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, cs4231_inb(chip, CS4231P(STATUS)));
@@ -452,10 +455,17 @@ static int snd_cs4231_trigger(snd_pcm_substream_t *substream,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		do_start = 1; break;
 	case SNDRV_PCM_TRIGGER_STOP:
-	{
-		unsigned int what = 0;
-		snd_pcm_substream_t *s = substream;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		do_start = 0; break;
+	default:
+		return -EINVAL;
+	}
+
+	what = 0;
+	s = substream;
 	do {
 		if (s == chip->playback_substream) {
 			what |= CS4231_PLAYBACK_ENABLE;
@@ -467,7 +477,7 @@ static int snd_cs4231_trigger(snd_pcm_substream_t *substream,
 		s = s->link_next;
 	} while (s != substream);
 	spin_lock(&chip->reg_lock);
-		if (cmd == SNDRV_PCM_TRIGGER_START) {
+	if (do_start) {
 		chip->image[CS4231_IFACE_CTRL] |= what;
 		if (chip->trigger)
 			chip->trigger(chip, what, 1);
@@ -478,12 +488,6 @@ static int snd_cs4231_trigger(snd_pcm_substream_t *substream,
 	}
 	snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]);
 	spin_unlock(&chip->reg_lock);
-		break;
-	}
-	default:
-		result = -EINVAL;
-		break;
-	}
 #if 0
 	snd_cs4231_debug(chip);
 #endif
@@ -1188,7 +1192,9 @@ int snd_cs4231_probe(cs4231_t *chip)
 static snd_pcm_hardware_t snd_cs4231_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
-				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_RESUME |
+				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
 				 SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
 	.rates =		SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,
@@ -1207,7 +1213,9 @@ static snd_pcm_hardware_t snd_cs4231_playback =
 static snd_pcm_hardware_t snd_cs4231_capture =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
-				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_RESUME |
+				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
 				 SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
 	.rates =		SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000,

sound/isa/es18xx.c

@@ -444,8 +444,7 @@ static int snd_es18xx_playback_hw_params(snd_pcm_substream_t * substream,
 	if (snd_pcm_format_width(params_format(hw_params)) == 16)
 		shift++;
 
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
 		if ((chip->caps & ES18XX_DUPLEX_MONO) &&
 		    (chip->capture_a_substream) &&
 		    params_channels(hw_params) != 1) {
@@ -453,10 +452,8 @@ static int snd_es18xx_playback_hw_params(snd_pcm_substream_t * substream,
 			return -EBUSY;
 		}
 		chip->dma2_shift = shift;
-		break;
-	case 1:
+	} else {
 		chip->dma1_shift = shift;
-		break;
 	}
 	if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0)
 		return err;
@@ -495,19 +492,12 @@ static int snd_es18xx_playback1_trigger(es18xx_t *chip,
 					snd_pcm_substream_t * substream,
 					int cmd)
 {
-        if (cmd == SNDRV_PCM_TRIGGER_START) {
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		if (chip->active & DAC2)
 			return 0;
 		chip->active |= DAC2;
-	} else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
-		if (!(chip->active & DAC2))
-			return 0;
-		chip->active &= ~DAC2;
-	} else {
-		return -EINVAL;
-	}
-
-	if (cmd == SNDRV_PCM_TRIGGER_START) {	
                 /* Start DMA */
 		if (chip->dma2 >= 4)
 			snd_es18xx_mixer_write(chip, 0x78, 0xb3);
@@ -523,8 +513,12 @@ static int snd_es18xx_playback1_trigger(es18xx_t *chip,
 			/* Enable PCM output */
 			snd_es18xx_dsp_command(chip, 0xD1);
 #endif
-        }
-        else {
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (!(chip->active & DAC2))
+			return 0;
+		chip->active &= ~DAC2;
                 /* Stop DMA */
                 snd_es18xx_mixer_write(chip, 0x78, 0x00);
 #ifdef AVOID_POPS
@@ -536,6 +530,9 @@ static int snd_es18xx_playback1_trigger(es18xx_t *chip,
 			/* Disable PCM output */
 			snd_es18xx_dsp_command(chip, 0xD3);
 #endif
+		break;
+	default:
+		return -EINVAL;
 	}
 
 	return 0;
@@ -608,24 +605,27 @@ static int snd_es18xx_capture_trigger(snd_pcm_substream_t *substream,
 {
         es18xx_t *chip = snd_pcm_substream_chip(substream);
 
-        if (cmd == SNDRV_PCM_TRIGGER_START) {
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		if (chip->active & ADC1)
 			return 0;
 		chip->active |= ADC1;
-	} else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+                /* Start DMA */
+                snd_es18xx_write(chip, 0xB8, 0x0f);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		if (!(chip->active & ADC1))
 			return 0;
 		chip->active &= ~ADC1;
-	} else {
+                /* Stop DMA */
+                snd_es18xx_write(chip, 0xB8, 0x00);
+		break;
+	default:
 		return -EINVAL;
 	}
 
-	if (cmd == SNDRV_PCM_TRIGGER_START)
-                /* Start DMA */
-                snd_es18xx_write(chip, 0xB8, 0x0f);
-        else
-                /* Stop DMA */
-                snd_es18xx_write(chip, 0xB8, 0x00);
 	return 0;
 }
 
@@ -670,19 +670,12 @@ static int snd_es18xx_playback2_trigger(es18xx_t *chip,
 					snd_pcm_substream_t *substream,
 					int cmd)
 {
-        if (cmd == SNDRV_PCM_TRIGGER_START) {
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		if (chip->active & DAC1)
 			return 0;
 		chip->active |= DAC1;
-        } else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
-		if (!(chip->active & DAC1))
-			return 0;
-		chip->active &= ~DAC1;
-	} else {
-		return -EINVAL;
-	}
-
-	if (cmd == SNDRV_PCM_TRIGGER_START) {
                 /* Start DMA */
                 snd_es18xx_write(chip, 0xB8, 0x05);
 #ifdef AVOID_POPS
@@ -691,8 +684,12 @@ static int snd_es18xx_playback2_trigger(es18xx_t *chip,
                 /* Enable Audio 1 */
                 snd_es18xx_dsp_command(chip, 0xD1);
 #endif
-        }
-        else {
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (!(chip->active & DAC1))
+			return 0;
+		chip->active &= ~DAC1;
                 /* Stop DMA */
                 snd_es18xx_write(chip, 0xB8, 0x00);
 #ifdef AVOID_POPS
@@ -701,33 +698,31 @@ static int snd_es18xx_playback2_trigger(es18xx_t *chip,
                 /* Disable Audio 1 */
                 snd_es18xx_dsp_command(chip, 0xD3);
 #endif
+		break;
+	default:
+		return -EINVAL;
 	}
+
 	return 0;
 }
 
 static int snd_es18xx_playback_prepare(snd_pcm_substream_t *substream)
 {
         es18xx_t *chip = snd_pcm_substream_chip(substream);
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
 		return snd_es18xx_playback1_prepare(chip, substream);
-	case 1:
+	else
 		return snd_es18xx_playback2_prepare(chip, substream);
-	}
-	return -EINVAL;
 }
 
 static int snd_es18xx_playback_trigger(snd_pcm_substream_t *substream,
 				       int cmd)
 {
         es18xx_t *chip = snd_pcm_substream_chip(substream);
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
 		return snd_es18xx_playback1_trigger(chip, substream, cmd);
-	case 1:
+	else
 		return snd_es18xx_playback2_trigger(chip, substream, cmd);
-	}
-	return -EINVAL;
 }
 
 static void snd_es18xx_interrupt(int irq, void *dev_id, struct pt_regs *regs)
@@ -798,19 +793,17 @@ static snd_pcm_uframes_t snd_es18xx_playback_pointer(snd_pcm_substream_t * subst
         es18xx_t *chip = snd_pcm_substream_chip(substream);
 	int pos;
 
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
 		if (!(chip->active & DAC2))
 			return 0;
 		pos = chip->dma2_size - snd_dma_residue(chip->dma2);
 		return pos >> chip->dma2_shift;
-	case 1:
+	} else {
 		if (!(chip->active & DAC1))
 			return 0;
 		pos = chip->dma1_size - snd_dma_residue(chip->dma1);
 		return pos >> chip->dma1_shift;
 	}
-	return 0;
 }
 
 static snd_pcm_uframes_t snd_es18xx_capture_pointer(snd_pcm_substream_t * substream)
@@ -827,6 +820,7 @@ static snd_pcm_uframes_t snd_es18xx_capture_pointer(snd_pcm_substream_t * substr
 static snd_pcm_hardware_t snd_es18xx_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_MMAP_VALID),
 	.formats =		(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | 
 				 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
@@ -846,6 +840,7 @@ static snd_pcm_hardware_t snd_es18xx_playback =
 static snd_pcm_hardware_t snd_es18xx_capture =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_MMAP_VALID),
 	.formats =		(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | 
 				 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE),
@@ -867,20 +862,17 @@ static int snd_es18xx_playback_open(snd_pcm_substream_t * substream)
 	snd_pcm_runtime_t *runtime = substream->runtime;
         es18xx_t *chip = snd_pcm_substream_chip(substream);
 
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) {
 		if ((chip->caps & ES18XX_DUPLEX_MONO) &&
 		    chip->capture_a_substream && 
 		    chip->capture_a_substream->runtime->channels != 1)
 			return -EAGAIN;
 		chip->playback_a_substream = substream;
-		break;
-	case 1:
+	} else if (substream->number <= 1) {
 		if (chip->capture_a_substream)
 			return -EAGAIN;
 		chip->playback_b_substream = substream;
-		break;
-	default:
+	} else {
 		snd_BUG();
 		return -EINVAL;
 	}
@@ -912,17 +904,10 @@ static int snd_es18xx_playback_close(snd_pcm_substream_t * substream)
 {
         es18xx_t *chip = snd_pcm_substream_chip(substream);
 
-	switch (substream->number) {
-	case 0:
+	if (substream->number == 0 && (chip->caps & ES18XX_PCM2))
 		chip->playback_a_substream = NULL;
-		break;
-	case 1:
+	else
 		chip->playback_b_substream = NULL;
-		break;
-	default:
-		snd_BUG();
-		return -EINVAL;
-	}
 	
 	snd_pcm_lib_free_pages(substream);
 	return 0;
@@ -1544,6 +1529,9 @@ static int __init snd_es18xx_probe(es18xx_t *chip)
 
         snd_printd("[0x%lx] ESS%x chip found\n", chip->port, chip->version);
 
+	if (chip->dma1 == chip->dma2)
+		chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME);
+
         return snd_es18xx_initialize(chip);
 }
 
@@ -1600,6 +1588,8 @@ int __init snd_es18xx_pcm(es18xx_t *chip, int device, snd_pcm_t ** rpcm)
         pcm->info_flags = 0;
 	if (chip->caps & ES18XX_DUPLEX_SAME)
 		pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX;
+	if (! (chip->caps & ES18XX_PCM2))
+		pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX;
 	sprintf(pcm->name, "ESS AudioDrive ES%x", chip->version);
         chip->pcm = pcm;
 
@@ -1709,7 +1699,7 @@ static int snd_es18xx_free(es18xx_t *chip)
 		disable_dma(chip->dma1);
 		free_dma(chip->dma1);
 	}
-	if (chip->dma2 >= 0) {
+	if (chip->dma2 >= 0 && chip->dma1 != chip->dma2) {
 		disable_dma(chip->dma2);
 		free_dma(chip->dma2);
 	}
@@ -1773,7 +1763,7 @@ static int __init snd_es18xx_new_device(snd_card_t * card,
 	}
 	chip->dma1 = dma1;
 
-	if (request_dma(dma2, "ES18xx DMA 2")) {
+	if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) {
 		snd_es18xx_free(chip);
 		printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2);
 		return -EBUSY;
@@ -2181,10 +2171,16 @@ static int __init snd_audiodrive_probe(int dev)
 #endif
 	sprintf(card->driver, "ES%x", chip->version);
 	sprintf(card->shortname, "ESS AudioDrive ES%x", chip->version);
+	if (xdma1 != xdma2)
 		sprintf(card->longname, "%s at 0x%lx, irq %d, dma1 %d, dma2 %d",
 			card->shortname,
 			chip->port,
 			xirq, xdma1, xdma2);
+	else
+		sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d",
+			card->shortname,
+			chip->port,
+			xirq, xdma1);
 	if ((err = snd_card_register(card)) < 0) {
 		snd_card_free(card);
 		return err;

sound/isa/sb/sb16.c

@@ -666,7 +666,7 @@ static int __init alsa_card_sb16_setup(char *str)
 {
 	static unsigned __initdata nr_dev = 0;
 	int __attribute__ ((__unused__)) pnp = INT_MAX;
-	int __attribute__ ((__unused__)) csp = INT_MAX;
+	int __attribute__ ((__unused__)) xcsp = INT_MAX;
 
 	if (nr_dev >= SNDRV_CARDS)
 		return 0;
@@ -683,7 +683,7 @@ static int __init alsa_card_sb16_setup(char *str)
 	       get_option(&str,&mic_agc[nr_dev]) == 2
 #ifdef CONFIG_SND_SB16_CSP
 	       &&
-	       get_option(&str,&csp[nr_dev]) == 2
+	       get_option(&str,&xcsp) == 2
 #endif
 #ifdef SNDRV_SBAWE_EMU8000
 	       &&
@@ -697,7 +697,7 @@ static int __init alsa_card_sb16_setup(char *str)
 #endif
 #ifdef CONFIG_SND_SB16_CSP
 	if (csp != INT_MAX)
-		csp[nr_dev] = csp;
+		csp[nr_dev] = xcsp;
 #endif
 	nr_dev++;
 	return 1;

sound/pci/ali5451/ali5451.c

@@ -1210,11 +1210,19 @@ static int snd_ali_trigger(snd_pcm_substream_t *substream,
 	unsigned int what, whati, capture_flag;
 	snd_ali_voice_t *pvoice = NULL, *evoice = NULL;
 	unsigned int val;
+	int do_start;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		do_start = 1; break;
 	case SNDRV_PCM_TRIGGER_STOP:
-	{
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		do_start = 0; break;
+	default:
+		return -EINVAL;
+	}
+
 	what = whati = capture_flag = 0;
 	s = substream;
 	do {
@@ -1228,10 +1236,14 @@ static int snd_ali_trigger(snd_pcm_substream_t *substream,
 				whati |= 1 << (evoice->number & 0x1f);
 				what |= 1 << (evoice->number & 0x1f);
 			}
-				if (cmd == SNDRV_PCM_TRIGGER_START) {
+			if (do_start) {
 				pvoice->running = 1;
 				if (evoice != NULL)
 					evoice->running = 1;
+			} else {
+				pvoice->running = 0;
+				if (evoice != NULL)
+					evoice->running = 0;
 			}
 			snd_pcm_trigger_done(s, substream);
 			if (pvoice->mode)
@@ -1240,29 +1252,22 @@ static int snd_ali_trigger(snd_pcm_substream_t *substream,
 		s = s->link_next;
 	} while (s != substream);
 	spin_lock(&codec->reg_lock);
-		if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+	if (! do_start) {
 		outl(what, ALI_REG(codec, ALI_STOP));
-			pvoice->running = 0;
-			if (evoice != NULL)
-				evoice->running = 0;
 	}
 	val = inl(ALI_REG(codec, ALI_AINTEN));
-		if (cmd == SNDRV_PCM_TRIGGER_START) {
+	if (do_start) {
 		val |= whati;
 	} else {
 		val &= ~whati;
 	}
 	outl(val, ALI_REG(codec, ALI_AINTEN));
-		if (cmd == SNDRV_PCM_TRIGGER_START) {
+	if (do_start) {
 		outl(what, ALI_REG(codec, ALI_START));
 	}
 	snd_ali_printk("trigger: what=%xh whati=%xh\n",what,whati);
 	spin_unlock(&codec->reg_lock);
-		break;
-	}
-	default:
-		return -EINVAL;
-	}
+
 	return 0;
 }
 
@@ -1544,7 +1549,9 @@ static snd_pcm_hardware_t snd_ali_playback =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_RESUME |
+				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
 				 SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
 	.rates =		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
@@ -1568,7 +1575,9 @@ static snd_pcm_hardware_t snd_ali_capture =
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-				 SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START),
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_RESUME |
+				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |
 				 SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U16_LE),
 	.rates =		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,

sound/pci/cs4281.c

@@ -735,12 +735,14 @@ static int snd_cs4281_trigger(snd_pcm_substream_t *substream, int cmd)
 		dma->valFCR &= ~BA0_FCR_FEN;
 		break;
 	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
 		snd_cs4281_pokeBA0(chip, dma->regDMR, dma->valDMR & ~BA0_DMR_DMA);
 		dma->valDMR |= BA0_DMR_DMA;
 		dma->valDCR &= ~BA0_DCR_MSK;
 		dma->valFCR |= BA0_FCR_FEN;
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 		dma->valDMR &= ~(BA0_DMR_DMA|BA0_DMR_POLL);
 		dma->valDCR |= BA0_DCR_MSK;
 		dma->valFCR &= ~BA0_FCR_FEN;
@@ -900,6 +902,7 @@ static snd_pcm_hardware_t snd_cs4281_playback =
 				 SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID |
 				 SNDRV_PCM_INFO_PAUSE |
+				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
 				SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |
@@ -925,6 +928,7 @@ static snd_pcm_hardware_t snd_cs4281_capture =
 				 SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID |
 				 SNDRV_PCM_INFO_PAUSE |
+				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 |
 				SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE |

sound/pci/cs46xx/cs46xx_lib.c

@@ -849,7 +849,7 @@ static snd_pcm_uframes_t snd_cs46xx_playback_indirect_pointer(snd_pcm_substream_
 	cpcm->hw_io = ptr;
 	cpcm->hw_ready -= bytes;
 	cpcm->sw_io += bytes;
-	if (cpcm->sw_io > cpcm->sw_bufsize)
+	if (cpcm->sw_io >= cpcm->sw_bufsize)
 		cpcm->sw_io -= cpcm->sw_bufsize;
 	snd_cs46xx_playback_transfer(substream, 0);
 	return cpcm->sw_io >> cpcm->shift;
@@ -874,7 +874,7 @@ static snd_pcm_uframes_t snd_cs46xx_capture_indirect_pointer(snd_pcm_substream_t
 	chip->capt.hw_io = ptr;
 	chip->capt.hw_ready += bytes;
 	chip->capt.sw_io += bytes;
-	if (chip->capt.sw_io > chip->capt.sw_bufsize)
+	if (chip->capt.sw_io >= chip->capt.sw_bufsize)
 		chip->capt.sw_io -= chip->capt.sw_bufsize;
 	snd_cs46xx_capture_transfer(substream, 0);
 	return chip->capt.sw_io >> chip->capt.shift;
@@ -952,7 +952,7 @@ static int snd_cs46xx_playback_trigger(snd_pcm_substream_t * substream,
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 		/* magic value to unmute PCM stream  playback volume */
 		snd_cs46xx_poke(chip, (cpcm->pcm_channel->pcm_reader_scb->address + 
-				       SCBVolumeCtrl) << 2, 0x80007fff);
+				       SCBVolumeCtrl) << 2, 0x80008000);
 
 		if (cpcm->pcm_channel->unlinked)
 			cs46xx_dsp_pcm_link(chip,cpcm->pcm_channel);
@@ -1032,32 +1032,27 @@ static int snd_cs46xx_capture_trigger(snd_pcm_substream_t * substream,
 static int _cs46xx_adjust_sample_rate (cs46xx_t *chip, cs46xx_pcm_t *cpcm,
 				       int sample_rate) 
 {
-	/* if this is the only PCMReaderSCB child under current
-	   SrcTask then there no need to re-create pcm-channel */
-	if ( cpcm->pcm_channel->src_scb->ref_count == 1 &&
-	     cpcm->pcm_channel->sample_rate != sample_rate &&
-	     /* never set a 0 khz sample rate */
-	     sample_rate) {
-		/* sample rate not set or we can reuse
-		   the same SRC*/
-		cs46xx_dsp_set_src_sample_rate (chip,cpcm->pcm_channel->src_scb,sample_rate);
-		cpcm->pcm_channel->sample_rate = sample_rate;
-	} 
 
-	/* if there is more then 1 PCMReaderSCB child's under current
-	   SrcTask then we must recreate channel */
-	if (cpcm->pcm_channel->sample_rate != sample_rate &&
-	    cpcm->pcm_channel->src_scb->ref_count != 1 &&
-	    /* never set a 0 khz sample rate */
-	    sample_rate) {
+	/* If PCMReaderSCB and SrcTaskSCB not created yet ... */
+	if ( cpcm->pcm_channel == NULL) {
+		cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, 
+								   cpcm, cpcm->hw_addr,cpcm->pcm_channel_id);
+		if (cpcm->pcm_channel == NULL) {
+			snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n");
+			return -ENOMEM;
+		}
+		cpcm->pcm_channel->sample_rate = sample_rate;
+	} else
+	/* if sample rate is changed */
+	if (cpcm->pcm_channel->sample_rate != sample_rate) {
 		int unlinked = cpcm->pcm_channel->unlinked;
 		cs46xx_dsp_destroy_pcm_channel (chip,cpcm->pcm_channel);
 
 		if ( (cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, sample_rate, cpcm, 
 									 cpcm->hw_addr,
-									 cpcm->pcm_channel->pcm_channel_id)) == NULL) {
+									 cpcm->pcm_channel_id)) == NULL) {
 			snd_printk(KERN_ERR "cs46xx: failed to re-create virtual PCM channel\n");
-			return -ENXIO;
+			return -ENOMEM;
 		}
 
 		if (!unlinked) cs46xx_dsp_pcm_link (chip,cpcm->pcm_channel);
@@ -1082,15 +1077,21 @@ static int snd_cs46xx_playback_hw_params(snd_pcm_substream_t * substream,
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-	down (&chip->spos_mutex);
+	snd_assert (sample_rate != 0, return -ENXIO);
 
-	snd_assert (cpcm->pcm_channel != NULL);
+	down (&chip->spos_mutex);
 
 	if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) {
 		up (&chip->spos_mutex);
 		return -ENXIO;
 	}
 
+	snd_assert (cpcm->pcm_channel != NULL);
+	if (!cpcm->pcm_channel) {
+		up (&chip->spos_mutex);
+		return -ENXIO;
+	}
+
 	if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size * 4)) {
 		 up (&chip->spos_mutex);
 		 return -EINVAL;
@@ -1108,11 +1109,11 @@ static int snd_cs46xx_playback_hw_params(snd_pcm_substream_t * substream,
 
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-		if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+		if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_rear_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_IEC958_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_iec958_ops;
 		} else {
 			snd_assert(0);
@@ -1135,11 +1136,11 @@ static int snd_cs46xx_playback_hw_params(snd_pcm_substream_t * substream,
 		}
 
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-		if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
+		if (cpcm->pcm_channel_id == DSP_PCM_MAIN_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_PCM_REAR_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_rear_ops;
-		} else if (cpcm->pcm_channel->pcm_channel_id == DSP_IEC958_CHANNEL) {
+		} else if (cpcm->pcm_channel_id == DSP_IEC958_CHANNEL) {
 			substream->ops = &snd_cs46xx_playback_indirect_iec958_ops;
 		} else {
 			snd_assert(0);
@@ -1165,6 +1166,10 @@ static int snd_cs46xx_playback_hw_free(snd_pcm_substream_t * substream)
 
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
+	/* if play_back open fails, then this function
+	   is called and cpcm can actually be NULL here */
+	if (!cpcm) return -ENXIO;
+
 	if (runtime->dma_area != cpcm->hw_area)
 		snd_pcm_lib_free_pages(substream);
     
@@ -1482,15 +1487,8 @@ static int _cs46xx_playback_open_channel (snd_pcm_substream_t * substream,int pc
 	cpcm->substream = substream;
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	down (&chip->spos_mutex);
-	cpcm->pcm_channel = cs46xx_dsp_create_pcm_channel (chip, runtime->rate, cpcm, cpcm->hw_addr,pcm_channel_id);
-
-	if (cpcm->pcm_channel == NULL) {
-		snd_printk(KERN_ERR "cs46xx: failed to create virtual PCM channel\n");
-		snd_free_pci_pages(chip->pci, cpcm->hw_size, cpcm->hw_area, cpcm->hw_addr);
-		snd_magic_kfree(cpcm);
-		up (&chip->spos_mutex);
-		return -ENOMEM;
-	}
+	cpcm->pcm_channel = NULL; 
+	cpcm->pcm_channel_id = pcm_channel_id;
 
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 
 				   &hw_constraints_period_sizes);
@@ -1583,6 +1581,9 @@ static int snd_cs46xx_playback_close(snd_pcm_substream_t * substream)
 
 	cpcm = snd_magic_cast(cs46xx_pcm_t, runtime->private_data, return -ENXIO);
 
+	/* when playback_open fails, then cpcm can be NULL */
+	if (!cpcm) return -ENXIO;
+
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	down (&chip->spos_mutex);
 	if (cpcm->pcm_channel) {
@@ -1887,8 +1888,8 @@ static int snd_cs46xx_vol_dac_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_
 {
 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
 
-	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_right;
-	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->dac_volume_left;
+	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->dac_volume_right;
 
 	return 0;
 }
@@ -1913,8 +1914,8 @@ static int snd_cs46xx_vol_iec958_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_val
 {
 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
 
-	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->spdif_input_volume_right;
-	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->spdif_input_volume_left;
+	ucontrol->value.integer.value[0] = chip->dsp_spos_instance->spdif_input_volume_left;
+	ucontrol->value.integer.value[1] = chip->dsp_spos_instance->spdif_input_volume_right;
 	return 0;
 }
 
@@ -1923,8 +1924,8 @@ static int snd_cs46xx_vol_iec958_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_val
 	cs46xx_t *chip = snd_kcontrol_chip(kcontrol);
 	int change = 0;
 
-	if (chip->dsp_spos_instance->spdif_input_volume_right != ucontrol->value.integer.value[0] ||
-	    chip->dsp_spos_instance->spdif_input_volume_left != ucontrol->value.integer.value[1]) {
+	if (chip->dsp_spos_instance->spdif_input_volume_left  != ucontrol->value.integer.value[0] ||
+	    chip->dsp_spos_instance->spdif_input_volume_right!= ucontrol->value.integer.value[1]) {
 		cs46xx_dsp_set_iec958_volume (chip,
 					      ucontrol->value.integer.value[0],
 					      ucontrol->value.integer.value[1]);
@@ -2206,6 +2207,7 @@ static int snd_cs46xx_spdif_stream_put(snd_kcontrol_t * kcontrol,
 
 #endif /* CONFIG_SND_CS46XX_NEW_DSP */
 
+
 #ifdef CONFIG_SND_CS46XX_DEBUG_GPIO
 static int snd_cs46xx_egpio_select_info(snd_kcontrol_t *kcontrol, 
                                         snd_ctl_elem_info_t *uinfo)
@@ -3331,8 +3333,8 @@ int __devinit snd_cs46xx_start_dsp(cs46xx_t *chip)
 	
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
 	/* set the attenuation to 0dB */ 
-	snd_cs46xx_poke(chip, (MASTERMIX_SCB_ADDR + 0xE) << 2, 0x80008000); 
-	snd_cs46xx_poke(chip, (VARIDECIMATE_SCB_ADDR + 0xE) << 2, 0x80008000);
+	/* snd_cs46xx_poke(chip, (MASTERMIX_SCB_ADDR + 0xE) << 2, 0x80008000); 
+	   snd_cs46xx_poke(chip, (VARIDECIMATE_SCB_ADDR + 0xE) << 2, 0x80008000); */
 
 	/*
 	 * Initialize cs46xx SPDIF controller

sound/pci/cs46xx/cs46xx_lib.h

@@ -152,6 +152,7 @@ dsp_scb_descriptor_t * cs46xx_dsp_create_pcm_reader_scb(cs46xx_t * chip,char * s
                                                         dsp_scb_descriptor_t * parent_scb,
                                                         int scb_child_type);
 dsp_scb_descriptor_t *  cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
+						       int sample_rate,
                                                        u16 src_buffer_addr,
                                                        u16 src_delay_buffer_addr,u32 dest,
                                                        dsp_scb_descriptor_t * parent_scb,
@@ -202,8 +203,6 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,u32 sa
                                                           int pcm_channel_id);
 void                       cs46xx_dsp_destroy_pcm_channel (cs46xx_t * chip,
                                                            pcm_channel_descriptor_t * pcm_channel);
-void                       cs46xx_dsp_set_src_sample_rate(cs46xx_t * chip,dsp_scb_descriptor_t * src, 
-                                                          u32 rate);
 int                        cs46xx_dsp_pcm_unlink (cs46xx_t * chip,pcm_channel_descriptor_t * pcm_channel);
 int                        cs46xx_dsp_pcm_link (cs46xx_t * chip,pcm_channel_descriptor_t * pcm_channel);
 dsp_scb_descriptor_t *     cs46xx_add_record_source (cs46xx_t *chip,dsp_scb_descriptor_t * source,

sound/pci/cs46xx/dsp_spos.c

@@ -615,7 +615,7 @@ static void cs46xx_dsp_proc_sample_dump_read (snd_info_entry_t *entry, snd_info_
 	snd_iprintf(buffer,"\nMIX_SAMPLE_BUF1:\n");
 
 	col = 0;
-	for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x30; i += sizeof(u32),col ++) {
+	for (i = MIX_SAMPLE_BUF1;i < MIX_SAMPLE_BUF1 + 0x40; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
@@ -628,9 +628,9 @@ static void cs46xx_dsp_proc_sample_dump_read (snd_info_entry_t *entry, snd_info_
 		snd_iprintf(buffer,"%08X ",readl(dst + i));
 	}
 
-	snd_iprintf(buffer,"\n\n");
+	snd_iprintf(buffer,"\nSRC_TASK_SCB1:\n");
 	col = 0;
-	for (i = SPDIFI_IP_OUTPUT_BUFFER1;i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x40; i += sizeof(u32),col ++) {
+	for (i = 0x2580 ; i < 0x2580 + 0x40 ; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
@@ -646,7 +646,7 @@ static void cs46xx_dsp_proc_sample_dump_read (snd_info_entry_t *entry, snd_info_
 
 	snd_iprintf(buffer,"\nSPDIFO_BUFFER:\n");
 	col = 0;
-	for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x40; i += sizeof(u32),col ++) {
+	for (i = SPDIFO_IP_OUTPUT_BUFFER1;i < SPDIFO_IP_OUTPUT_BUFFER1 + 0x30; i += sizeof(u32),col ++) {
 		if (col == 4) {
 			snd_iprintf(buffer,"\n");
 			col = 0;
@@ -705,7 +705,7 @@ static void cs46xx_dsp_proc_sample_dump_read (snd_info_entry_t *entry, snd_info_
 
 		snd_iprintf(buffer,"%08X ",readl(dst + i));
 	}
-
+#if 0
 	snd_iprintf(buffer,"\nWRITE_BACK_BUF1: \n");
 	col = 0;
 	for (i = WRITE_BACK_BUF1;i < WRITE_BACK_BUF1 + 0x40; i += sizeof(u32),col ++) {
@@ -720,7 +720,22 @@ static void cs46xx_dsp_proc_sample_dump_read (snd_info_entry_t *entry, snd_info_
 
 		snd_iprintf(buffer,"%08X ",readl(dst + i));
 	}
+#endif
+
+	snd_iprintf(buffer,"\nSPDIFI_IP_OUTPUT_BUFFER1: \n");
+	col = 0;
+	for (i = SPDIFI_IP_OUTPUT_BUFFER1;i < SPDIFI_IP_OUTPUT_BUFFER1 + 0x80; i += sizeof(u32),col ++) {
+		if (col == 4) {
+			snd_iprintf(buffer,"\n");
+			col = 0;
+		}
+
+		if (col == 0) {
+			snd_iprintf(buffer, "%04X ",i);
+		}
 		
+		snd_iprintf(buffer,"%08X ",readl(dst + i));
+	}
 	snd_iprintf(buffer,"\n");
 }
 
@@ -1356,6 +1371,7 @@ int cs46xx_dsp_scb_and_task_init (cs46xx_t *chip)
 
 	/* SPDIF input sampel rate converter */
 	src_task_scb = cs46xx_dsp_create_src_task_scb(chip,"SrcTaskSCB_SPDIFI",
+						      48000,
 						      SRC_OUTPUT_BUF1,
 						      SRC_DELAY_BUF1,SRCTASK_SCB_ADDR,
 						      master_mix_scb,
@@ -1597,11 +1613,14 @@ int cs46xx_dsp_enable_spdif_in (cs46xx_t *chip)
 	cs46xx_poke_via_dsp (chip,SP_SPDIN_FIFOPTR, 0x0);
 	cs46xx_src_link(chip,ins->spdif_in_src);
 
+	/* unmute SRC volume */
+	cs46xx_dsp_scb_set_volume (chip,ins->spdif_in_src,0x7fff,0x7fff);
+
 	spin_unlock_irq(&chip->reg_lock);
 
 	/* set SPDIF input sample rate and unmute
 	   NOTE: only 48khz support for SPDIF input this time */
-	cs46xx_dsp_set_src_sample_rate(chip,ins->spdif_in_src,48000);
+	/* cs46xx_dsp_set_src_sample_rate(chip,ins->spdif_in_src,48000); */
 
 	/* monitor state */
 	ins->spdif_status_in = 1;

sound/pci/cs46xx/dsp_spos.h

@@ -197,7 +197,7 @@ static inline u8 _wrap_all_bits (u8 val) {
 		((val & 0x10) >> 1) |
 		((val & 0x20) >> 3) |
 		((val & 0x40) >> 5) |
-		((val & 0x80) >> 6);
+		((val & 0x80) >> 7);
 
 	return wrapped;
 

sound/pci/cs46xx/dsp_spos_scb_lib.c

@@ -576,8 +576,11 @@ cs46xx_dsp_create_pcm_reader_scb(cs46xx_t * chip,char * scb_name,
 	return scb;
 }
 
+#define GOF_PER_SEC 200
+
 dsp_scb_descriptor_t * 
 cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
+			       int rate,
                                u16 src_buffer_addr,
                                u16 src_delay_buffer_addr,u32 dest,
                                dsp_scb_descriptor_t * parent_scb,
@@ -586,22 +589,57 @@ cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
 
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 	dsp_scb_descriptor_t * scb;
+	unsigned int tmp1, tmp2;
+	unsigned int phiIncr;
+	unsigned int correctionPerGOF, correctionPerSec;
+
+	snd_printdd( "dsp_spos: setting %s rate to %u\n",scb_name,rate);
 
+	/*
+	 *  Compute the values used to drive the actual sample rate conversion.
+	 *  The following formulas are being computed, using inline assembly
+	 *  since we need to use 64 bit arithmetic to compute the values:
+	 *
+	 *  phiIncr = floor((Fs,in * 2^26) / Fs,out)
+	 *  correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
+	 *                                   GOF_PER_SEC)
+	 *  ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
+	 *                       GOF_PER_SEC * correctionPerGOF
+	 *
+	 *  i.e.
+	 *
+	 *  phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
+	 *  correctionPerGOF:correctionPerSec =
+	 *      dividend:remainder(ulOther / GOF_PER_SEC)
+	 */
+	tmp1 = rate << 16;
+	phiIncr = tmp1 / 48000;
+	tmp1 -= phiIncr * 48000;
+	tmp1 <<= 10;
+	phiIncr <<= 10;
+	tmp2 = tmp1 / 48000;
+	phiIncr += tmp2;
+	tmp1 -= tmp2 * 48000;
+	correctionPerGOF = tmp1 / GOF_PER_SEC;
+	tmp1 -= correctionPerGOF * GOF_PER_SEC;
+	correctionPerSec = tmp1;
+
+	{
 		src_task_scb_t src_task_scb = {
 			0x0028,0x00c8,
 			0x5555,0x0000,
 			0x0000,0x0000,
 			src_buffer_addr,1,
-		0x0028,0x00c8,
+			correctionPerGOF,correctionPerSec,
 			RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_32,  
 			0x0000,src_delay_buffer_addr,                  
 			0x0,                                            
-		0x80,(src_delay_buffer_addr + (24 * 4)),
+			0x080,(src_delay_buffer_addr + (24 * 4)),
 			0,0, /* next_scb, sub_list_ptr */
 			0,0, /* entry, this_spb */
 			RSCONFIG_SAMPLE_16STEREO + RSCONFIG_MODULO_8,
 			src_buffer_addr << 0x10,
-		0x04000000,
+			phiIncr,
 			{ 
 				0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left,
 				0xffff - ins->dac_volume_right,0xffff - ins->dac_volume_left
@@ -625,6 +663,7 @@ cs46xx_dsp_create_src_task_scb(cs46xx_t * chip,char * scb_name,
 		scb = _dsp_create_generic_scb(chip,scb_name,(u32 *)&src_task_scb,
 					      dest,ins->s16_up,parent_scb,
 					      scb_child_type);
+	}
 
 	return scb;
 }
@@ -1093,6 +1132,8 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 {
 	dsp_spos_instance_t * ins = chip->dsp_spos_instance;
 	dsp_scb_descriptor_t * src_scb = NULL,* pcm_scb, * mixer_scb = NULL;
+	dsp_scb_descriptor_t * src_parent_scb = NULL;
+
 	/*dsp_scb_descriptor_t * pcm_parent_scb;*/
 	char scb_name[DSP_MAX_SCB_NAME];
 	int i,pcm_index = -1, insert_point, src_index = -1;
@@ -1116,13 +1157,14 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 	case DSP_IEC958_CHANNEL:
 		snd_assert (ins->asynch_tx_scb != NULL, return NULL);
 		mixer_scb = ins->asynch_tx_scb;
-#if 0
-		if (ins->spdif_status_out & DSP_SPDIF_STATUS_AC3_MODE) {
-			snd_printdd ("IEC958 opened in AC3 mode\n");
-			/*src_scb = ins->asynch_tx_scb;
-			  ins->asynch_tx_scb->ref_count ++;*/
+
+		/* if sample rate is set to 48khz we pass
+		   the Sample Rate Converted (which could
+		   alter the raw data stream ...) */
+		if (sample_rate == 48000) {
+			snd_printdd ("IEC958 pass through\n");
+			src_parent_scb = ins->asynch_tx_scb;
 		}
-#endif
 		break;
 	default:
 		snd_assert (0);
@@ -1158,8 +1200,6 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 	}
 
 	if (src_scb == NULL) {
-		dsp_scb_descriptor_t * src_parent_scb;
-
 		if (ins->nsrc_scb >= DSP_MAX_SRC_NR) {
 			snd_printk(KERN_ERR "dsp_spos: to many SRC instances\n!");
 			return NULL;
@@ -1176,6 +1216,7 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 		snd_assert (src_index != -1,return NULL);
 
 		/* we need to create a new SRC SCB */
+		if (src_parent_scb == NULL) {
 			if (mixer_scb->sub_list_ptr == ins->the_null_scb) {
 				src_parent_scb = mixer_scb;
 				insert_point = SCB_ON_PARENT_SUBLIST_SCB;
@@ -1183,11 +1224,13 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 				src_parent_scb = find_next_free_scb(chip,mixer_scb->sub_list_ptr);
 				insert_point = SCB_ON_PARENT_NEXT_SCB;
 			}
+		} else insert_point = SCB_ON_PARENT_NEXT_SCB;
 
 		snprintf (scb_name,DSP_MAX_SCB_NAME,"SrcTask_SCB%d",src_index);
 
 		snd_printdd( "dsp_spos: creating SRC \"%s\"\n",scb_name);
 		src_scb = cs46xx_dsp_create_src_task_scb(chip,scb_name,
+							 sample_rate,
 							 src_output_buffer_addr[src_index],
 							 src_delay_buffer_addr[src_index],
 							 /* 0x400 - 0x600 source SCBs */
@@ -1200,7 +1243,7 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 			return NULL;
 		}
 
-		cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate);
+		/* cs46xx_dsp_set_src_sample_rate(chip,src_scb,sample_rate); */
 
 		ins->nsrc_scb ++;
 	} 
@@ -1226,6 +1269,19 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 		return NULL;
 	}
 
+	if (pcm_channel_id == DSP_IEC958_CHANNEL && sample_rate == 48000) {
+		snd_assert (ins->spdif_pcm_input_scb == NULL);
+		
+		/* a hack to make the skip the SRC and pass the stream 
+		   directly to the SPDIF task */
+		ins->spdif_pcm_input_scb = 
+			cs46xx_dsp_create_pcm_serial_input_scb(chip,"PCMSerialInput_PCM",
+							       PCMSERIALINII_SCB_ADDR,
+							       pcm_scb,
+							       ins->asynch_tx_scb,
+							       SCB_ON_PARENT_SUBLIST_SCB);		
+	}
+	
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	ins->pcm_channels[pcm_index].sample_rate = sample_rate;
 	ins->pcm_channels[pcm_index].pcm_reader_scb = pcm_scb;
@@ -1236,7 +1292,6 @@ pcm_channel_descriptor_t * cs46xx_dsp_create_pcm_channel (cs46xx_t * chip,
 	ins->pcm_channels[pcm_index].active = 1;
 	ins->pcm_channels[pcm_index].pcm_slot = pcm_index;
 	ins->pcm_channels[pcm_index].mixer_scb = mixer_scb;
-	ins->pcm_channels[pcm_index].pcm_channel_id = pcm_channel_id;
 	ins->npcm_channels ++;
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 
@@ -1416,59 +1471,6 @@ int cs46xx_dsp_pcm_link (cs46xx_t * chip,pcm_channel_descriptor_t * pcm_channel)
 	return 0;
 }
 
-
-#define GOF_PER_SEC 200
-  
-void cs46xx_dsp_set_src_sample_rate(cs46xx_t *chip,dsp_scb_descriptor_t * src, u32 rate)
-{
-	unsigned long flags;
-	unsigned int tmp1, tmp2;
-	unsigned int phiIncr;
-	unsigned int correctionPerGOF, correctionPerSec;
-
-	snd_printdd( "dsp_spos: setting SRC rate to %u\n",rate);
-	/*
-	 *  Compute the values used to drive the actual sample rate conversion.
-	 *  The following formulas are being computed, using inline assembly
-	 *  since we need to use 64 bit arithmetic to compute the values:
-	 *
-	 *  phiIncr = floor((Fs,in * 2^26) / Fs,out)
-	 *  correctionPerGOF = floor((Fs,in * 2^26 - Fs,out * phiIncr) /
-	 *                                   GOF_PER_SEC)
-	 *  ulCorrectionPerSec = Fs,in * 2^26 - Fs,out * phiIncr -M
-	 *                       GOF_PER_SEC * correctionPerGOF
-	 *
-	 *  i.e.
-	 *
-	 *  phiIncr:other = dividend:remainder((Fs,in * 2^26) / Fs,out)
-	 *  correctionPerGOF:correctionPerSec =
-	 *      dividend:remainder(ulOther / GOF_PER_SEC)
-	 */
-	tmp1 = rate << 16;
-	phiIncr = tmp1 / 48000;
-	tmp1 -= phiIncr * 48000;
-	tmp1 <<= 10;
-	phiIncr <<= 10;
-	tmp2 = tmp1 / 48000;
-	phiIncr += tmp2;
-	tmp1 -= tmp2 * 48000;
-	correctionPerGOF = tmp1 / GOF_PER_SEC;
-	tmp1 -= correctionPerGOF * GOF_PER_SEC;
-	correctionPerSec = tmp1;
-
-	/*
-	 *  Fill in the SampleRateConverter control block.
-	 */
-	spin_lock_irqsave(&chip->reg_lock, flags);
-
-	snd_cs46xx_poke(chip, (src->address + SRCCorPerGof) << 2,
-	  ((correctionPerSec << 16) & 0xFFFF0000) | (correctionPerGOF & 0xFFFF));
-
-	snd_cs46xx_poke(chip, (src->address + SRCPhiIncr6Int26Frac) << 2, phiIncr);
-
-	spin_unlock_irqrestore(&chip->reg_lock, flags);
-}
-
 dsp_scb_descriptor_t * cs46xx_add_record_source (cs46xx_t *chip,dsp_scb_descriptor_t * source,
                                                  u16 addr,char * scb_name)
 {
@@ -1655,10 +1657,19 @@ int cs46xx_iec958_post_close (cs46xx_t *chip)
 	cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV, ins->spdif_csuv_default);
 	
 	/* deallocate stuff */
+	if (ins->spdif_pcm_input_scb != NULL) {
+		cs46xx_dsp_remove_scb (chip,ins->spdif_pcm_input_scb);
+		ins->spdif_pcm_input_scb = NULL;
+	}
+
 	cs46xx_dsp_remove_scb (chip,ins->asynch_tx_scb);
 	ins->asynch_tx_scb = NULL;
 
+	/* clear buffer to prevent any undesired noise */
+	_dsp_clear_sample_buffer(chip,SPDIFO_IP_OUTPUT_BUFFER1,256);
+
 	/* restore state */
+
 	if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
 		cs46xx_dsp_enable_spdif_out (chip);
 	}

sound/pci/emu10k1/emu10k1.c

@@ -199,7 +199,7 @@ static void __devexit snd_card_emu10k1_remove(struct pci_dev *pci)
 }
 
 static struct pci_driver driver = {
-	.name = "EMU10K1/Audigy",
+	.name = "EMU10K1_Audigy",
 	.id_table = snd_emu10k1_ids,
 	.probe = snd_card_emu10k1_probe,
 	.remove = __devexit_p(snd_card_emu10k1_remove),

sound/pci/ice1712/ak4524.c

@@ -217,9 +217,11 @@ void __devinit snd_ice1712_ak4524_init(ice1712_t *ice)
 
 #define AK_GET_CHIP(val)		(((val) >> 8) & 0xff)
 #define AK_GET_ADDR(val)		((val) & 0xff)
-#define AK_GET_SHIFT(val)		(((val) >> 16) & 0xff)
+#define AK_GET_SHIFT(val)		(((val) >> 16) & 0x7f)
+#define AK_GET_INVERT(val)		(((val) >> 23) & 1)
 #define AK_GET_MASK(val)		(((val) >> 24) & 0xff)
 #define AK_COMPOSE(chip,addr,shift,mask) (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24))
+#define AK_INVERT 			(1<<23)
 
 static int snd_ice1712_ak4524_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
 {
@@ -237,8 +239,11 @@ static int snd_ice1712_ak4524_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_
 	ice1712_t *ice = snd_kcontrol_chip(kcontrol);
 	int chip = AK_GET_CHIP(kcontrol->private_value);
 	int addr = AK_GET_ADDR(kcontrol->private_value);
+	int invert = AK_GET_INVERT(kcontrol->private_value);
 	unsigned int mask = AK_GET_MASK(kcontrol->private_value);
-	ucontrol->value.integer.value[0] = mask - ice->ak4524.images[chip][addr];
+	unsigned char val = ice->ak4524.images[chip][addr];
+	
+	ucontrol->value.integer.value[0] = invert ? mask - val : val;
 	return 0;
 }
 
@@ -247,9 +252,14 @@ static int snd_ice1712_ak4524_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_
 	ice1712_t *ice = snd_kcontrol_chip(kcontrol);
 	int chip = AK_GET_CHIP(kcontrol->private_value);
 	int addr = AK_GET_ADDR(kcontrol->private_value);
+	int invert = AK_GET_INVERT(kcontrol->private_value);
 	unsigned int mask = AK_GET_MASK(kcontrol->private_value);
-	unsigned char nval = mask - (ucontrol->value.integer.value[0] % (mask+1));
-	int change = ice->ak4524.images[chip][addr] != nval;
+	unsigned char nval = ucontrol->value.integer.value[0] % (mask+1);
+	int change;
+
+	if (invert)
+		nval = mask - nval;
+	change = ice->ak4524.images[chip][addr] != nval;
 	if (change)
 		snd_ice1712_ak4524_write(ice, chip, addr, nval);
 	return change;
@@ -353,7 +363,7 @@ int __devinit snd_ice1712_ak4524_build_controls(ice1712_t *ice)
 			break;
 		case SND_AK4529: {
 			int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb; /* registers 2-7 and b,c */
-			ctl.private_value = AK_COMPOSE(0, val, 0, 255);
+			ctl.private_value = AK_COMPOSE(0, val, 0, 255) | AK_INVERT;
 			break;
 		}
 		}

sound/pci/ice1712/hoontech.c

@@ -218,6 +218,11 @@ struct snd_ice1712_card_info snd_ice1712_hoontech_cards[] __devinitdata = {
 		"Hoontech SoundTrack Audio DSP24",
 		snd_ice1712_hoontech_init,
 	},
+	{
+		ICE1712_SUBDEVICE_STDSP24_MEDIA7_1,
+		"Hoontech STA DSP24 Media 7.1",
+		snd_ice1712_hoontech_init,
+	},
 	{ } /* terminator */
 };
 

sound/pci/ice1712/hoontech.h

@@ -27,6 +27,7 @@
 #define  HOONTECH_DEVICE_DESC 	       "{Hoontech SoundTrack DSP 24},"
 
 #define ICE1712_SUBDEVICE_STDSP24	0x12141217	/* Hoontech SoundTrack Audio DSP 24 */
+#define ICE1712_SUBDEVICE_STDSP24_MEDIA7_1	0x16141217	/* Hoontech ST Audio DSP24 Media 7.1 */
 
 extern struct snd_ice1712_card_info snd_ice1712_hoontech_cards[];
 

sound/pci/rme32.c

@@ -23,9 +23,38 @@
  *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
  * 
  * 
- *   ToDo: full duplex (32, 32/8, 32Pro)
+ * ****************************************************************************
+ * 
+ * Note #1 "Sek'd models" ................................... martin 2002-12-07
+ * 
+ * Identical soundcards by Sek'd were labeled:
+ * RME Digi 32     = Sek'd Prodif 32
+ * RME Digi 32 Pro = Sek'd Prodif 96
+ * RME Digi 32/8   = Sek'd Prodif Gold
+ * 
+ * ****************************************************************************
+ * 
+ * Note #2 "full duplex mode" ............................... martin 2002-12-07
+ * 
+ * Full duplex doesn't work. All cards (32, 32/8, 32Pro) are working identical
+ * in this mode. Rec data and play data are using the same buffer therefore. At
+ * first you have got the playing bits in the buffer and then (after playing
+ * them) they were overwitten by the captured sound of the CS8412/14. Both 
+ * modes (play/record) are running harmonically hand in hand in the same buffer
+ * and you have only one start bit plus one interrupt bit to control this 
+ * paired action.
+ * This is opposite to the latter rme96 where playing and capturing is totally
+ * separated and so their full duplex mode is supported by alsa (using two 
+ * start bits and two interrupts for two different buffers). 
+ * But due to the wrong sequence of playing and capturing ALSA shows no solved
+ * full duplex support for the rme32 at the moment. That's bad, but I'm not
+ * able to solve it. Are you motivated enough to solve this problem now? Your
+ * patch would be welcome!
+ * 
+ * ****************************************************************************
  */
 
+
 #include <sound/driver.h>
 #include <linux/delay.h>
 #include <linux/init.h>
@@ -80,29 +109,29 @@ MODULE_DEVICES("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}");
 #define RME32_IO_RESET_POS          0x20100
 
 /* Write control register bits */
-#define RME32_WCR_START     (1 << 0)
-#define RME32_WCR_MONO      (1 << 1)    /* 0: stereo, 1: mono
+#define RME32_WCR_START     (1 << 0)    /* startbit */
+#define RME32_WCR_MONO      (1 << 1)    /* 0=stereo, 1=mono
                                            Setting the whole card to mono
-                                           don't seems to be very useful.
+                                           doesn't seem to be very useful.
                                            A software-solution can handle 
                                            full-duplex with one direction in
                                            stereo and the other way in mono. 
                                            So, the hardware should work all 
                                            the time in stereo! */
-#define RME32_WCR_MODE24    (1 << 2)
-#define RME32_WCR_SEL       (1 << 3)
-#define RME32_WCR_FREQ_0    (1 << 4)
+#define RME32_WCR_MODE24    (1 << 2)    /* 0=16bit, 1=32bit */
+#define RME32_WCR_SEL       (1 << 3)    /* 0=input on output, 1=normal playback/capture */
+#define RME32_WCR_FREQ_0    (1 << 4)    /* frequency (play) */
 #define RME32_WCR_FREQ_1    (1 << 5)
-#define RME32_WCR_INP_0     (1 << 6)
+#define RME32_WCR_INP_0     (1 << 6)    /* input switch */
 #define RME32_WCR_INP_1     (1 << 7)
-#define RME32_WCR_RESET     (1 << 8)
-#define RME32_WCR_MUTE      (1 << 9)
-#define RME32_WCR_PRO       (1 << 10)
-#define RME32_WCR_DS_BM     (1 << 11)	/* only PRO/Adat-Version */
-#define RME32_WCR_ADAT      (1 << 12)	/* only Adat-Version */
-#define RME32_WCR_AUTOSYNC  (1 << 13)
-#define RME32_WCR_PD        (1 << 14)	/* only PRO-Version */
-#define RME32_WCR_EMP       (1 << 15)	/* only PRO-Version */
+#define RME32_WCR_RESET     (1 << 8)    /* Reset address */
+#define RME32_WCR_MUTE      (1 << 9)    /* digital mute for output */
+#define RME32_WCR_PRO       (1 << 10)   /* 1=professional, 0=consumer */
+#define RME32_WCR_DS_BM     (1 << 11)	/* 1=DoubleSpeed (only PRO-Version); 1=BlockMode (only Adat-Version) */
+#define RME32_WCR_ADAT      (1 << 12)	/* Adat Mode (only Adat-Version) */
+#define RME32_WCR_AUTOSYNC  (1 << 13)   /* AutoSync */
+#define RME32_WCR_PD        (1 << 14)	/* DAC Reset (only PRO-Version) */
+#define RME32_WCR_EMP       (1 << 15)	/* 1=Emphasis on (only PRO-Version) */
 
 #define RME32_WCR_BITPOS_FREQ_0 4
 #define RME32_WCR_BITPOS_FREQ_1 5
@@ -111,13 +140,13 @@ MODULE_DEVICES("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}");
 
 /* Read control register bits */
 #define RME32_RCR_AUDIO_ADDR_MASK 0x10001
-#define RME32_RCR_LOCK      (1 << 23)
-#define RME32_RCR_ERF       (1 << 26)
-#define RME32_RCR_FREQ_0    (1 << 27)
+#define RME32_RCR_LOCK      (1 << 23)   /* 1=locked, 0=not locked */
+#define RME32_RCR_ERF       (1 << 26)   /* 1=Error, 0=no Error */
+#define RME32_RCR_FREQ_0    (1 << 27)   /* CS841x frequency (record) */
 #define RME32_RCR_FREQ_1    (1 << 28)
 #define RME32_RCR_FREQ_2    (1 << 29)
-#define RME32_RCR_KMODE     (1 << 30)
-#define RME32_RCR_IRQ       (1 << 31)
+#define RME32_RCR_KMODE     (1 << 30)   /* card mode: 1=PLL, 0=quartz */
+#define RME32_RCR_IRQ       (1 << 31)   /* interrupt */
 
 #define RME32_RCR_BITPOS_F0 27
 #define RME32_RCR_BITPOS_F1 28
@@ -153,12 +182,12 @@ MODULE_DEVICES("{{RME,Digi32}," "{RME,Digi32/8}," "{RME,Digi32 PRO}}");
 #ifndef PCI_DEVICE_ID_DIGI32
 # define PCI_DEVICE_ID_DIGI32 0x9896
 #endif
-#ifndef PCI_DEVICE_ID_DIGI32_8
-# define PCI_DEVICE_ID_DIGI32_8 0x9898
-#endif
 #ifndef PCI_DEVICE_ID_DIGI32_PRO
 # define PCI_DEVICE_ID_DIGI32_PRO 0x9897
 #endif
+#ifndef PCI_DEVICE_ID_DIGI32_8
+# define PCI_DEVICE_ID_DIGI32_8 0x9898
+#endif
 
 typedef struct snd_rme32 {
 	spinlock_t lock;
@@ -314,10 +343,13 @@ static snd_pcm_hardware_t snd_rme32_playback_spdif_info = {
 static snd_pcm_hardware_t snd_rme32_capture_spdif_info = {
 	.info =		(SNDRV_PCM_INFO_MMAP |
 			 SNDRV_PCM_INFO_MMAP_VALID |
-			 SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE),
-	.formats =	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE),
+			 SNDRV_PCM_INFO_INTERLEAVED | 
+			 SNDRV_PCM_INFO_PAUSE),
+	.formats =	(SNDRV_PCM_FMTBIT_S16_LE | 
+			 SNDRV_PCM_FMTBIT_S32_LE),
 	.rates =	(SNDRV_PCM_RATE_32000 |
-			 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000),
+			 SNDRV_PCM_RATE_44100 | 
+			 SNDRV_PCM_RATE_48000),
 	.rate_min =	32000,
 	.rate_max =	48000,
 	.channels_min =	2,

sound/usb/Kconfig

@@ -5,7 +5,7 @@ menu "ALSA USB devices"
 
 config SND_USB_AUDIO
 	tristate "USB Audio/MIDI driver"
-	depends on SND
+	depends on SND && USB
 	help
 	  Say 'Y' or 'M' to include support for USB audio and USB MIDI devices.
 

sound/usb/usbaudio.h

@@ -172,7 +172,8 @@ struct snd_usb_midi_endpoint_info {
 
 /* for QUIRK_MIDI_YAMAHA, data is NULL */
 
-/* for QUIRK_MIDI_MIDIMAN, data is the number of ports */
+/* for QUIRK_MIDI_MIDIMAN, data points to a snd_usb_midi_endpoint_info
+ * structure (out_cables and in_cables only) */
 
 /* for QUIRK_ROLAND_UA100, data is NULL */
 

sound/usb/usbmidi.c

@@ -172,7 +172,7 @@ static void snd_usbmidi_input_packet(snd_usb_midi_in_endpoint_t* ep,
 /*
  * Processes the data read from the device.
  */
-static void snd_usbmidi_in_urb_complete(struct urb* urb, struct pt_regs *regs)
+static void snd_usbmidi_in_urb_complete(struct urb* urb)
 {
 	snd_usb_midi_in_endpoint_t* ep = snd_magic_cast(snd_usb_midi_in_endpoint_t, urb->context, return);
 
@@ -197,7 +197,7 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb, struct pt_regs *regs)
 /*
  * Converts the data read from a Midiman device to standard USB MIDI packets.
  */
-static void snd_usbmidi_in_midiman_complete(struct urb* urb, struct pt_regs *regs)
+static void snd_usbmidi_in_midiman_complete(struct urb* urb)
 {
 	if (urb->status == 0) {
 		uint8_t* buffer = (uint8_t*)urb->transfer_buffer;
@@ -223,10 +223,10 @@ static void snd_usbmidi_in_midiman_complete(struct urb* urb, struct pt_regs *reg
 			}
 		}
 	}
-	snd_usbmidi_in_urb_complete(urb, regs);
+	snd_usbmidi_in_urb_complete(urb);
 }
 
-static void snd_usbmidi_out_urb_complete(struct urb* urb, struct pt_regs *regs)
+static void snd_usbmidi_out_urb_complete(struct urb* urb)
 {
 	snd_usb_midi_out_endpoint_t* ep = snd_magic_cast(snd_usb_midi_out_endpoint_t, urb->context, return);
 
@@ -892,7 +892,8 @@ static int snd_usbmidi_detect_yamaha(snd_usb_midi_t* umidi,
 /*
  * Creates the endpoints and their ports for Midiman devices.
  */
-static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi, int ports)
+static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi,
+						snd_usb_midi_endpoint_info_t* endpoint)
 {
 	snd_usb_midi_endpoint_info_t ep_info;
 	struct usb_interface* intf;
@@ -906,7 +907,7 @@ static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi, int ports
 		return -ENOENT;
 	hostif = intf->altsetting;
 	intfd = get_iface_desc(hostif);
-	if (intfd->bNumEndpoints < (ports > 1 ? 5 : 3)) {
+	if (intfd->bNumEndpoints < (endpoint->out_cables > 0x0001 ? 5 : 3)) {
 		snd_printdd(KERN_ERR "not enough endpoints\n");
 		return -ENOENT;
 	}
@@ -923,7 +924,7 @@ static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi, int ports
 		snd_printdd(KERN_ERR "endpoint[2] isn't bulk output\n");
 		return -ENXIO;
 	}
-	if (ports > 1) {
+	if (endpoint->out_cables > 0x0001) {
 		epd = get_endpoint(hostif, 4);
 		if ((epd->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_OUT ||
 		    (epd->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_BULK) {
@@ -933,29 +934,31 @@ static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi, int ports
 	}
 
 	ep_info.epnum = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-	ep_info.out_cables = 0x5555 & ((1 << ports) - 1);
+	ep_info.out_cables = endpoint->out_cables & 0x5555;
 	err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
 	if (err < 0)
 		return err;
 
 	ep_info.epnum = get_endpoint(hostif, 0)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-	ep_info.in_cables = (1 << ports) - 1;
+	ep_info.in_cables = endpoint->in_cables;
 	err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
 	if (err < 0)
 		return err;
 	umidi->endpoints[0].in->urb->complete = snd_usbmidi_in_midiman_complete;
 
-	if (ports > 1) {
+	if (endpoint->out_cables > 0x0001) {
 		ep_info.epnum = get_endpoint(hostif, 4)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
-		ep_info.out_cables = 0xaaaa & ((1 << ports) - 1);
+		ep_info.out_cables = endpoint->out_cables & 0xaaaa;
 		err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[1]);
 		if (err < 0)
 			return err;
 	}
 
-	for (cable = 0; cable < ports; ++cable) {
+	for (cable = 0; cable < 0x10; ++cable) {
+		if (endpoint->out_cables & (1 << cable))
 			snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_OUTPUT, cable,
 						   &umidi->endpoints[cable & 1].out->ports[cable].substream);
+		if (endpoint->in_cables & (1 << cable))
 			snd_usbmidi_init_substream(umidi, SNDRV_RAWMIDI_STREAM_INPUT, cable,
 						   &umidi->endpoints[0].in->ports[cable].substream);
 	}
@@ -1020,6 +1023,8 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip,
 			err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]);
 			break;
 		case QUIRK_MIDI_MIDIMAN:
+			memcpy(&endpoints[0], quirk->data,
+			       sizeof(snd_usb_midi_endpoint_info_t));
 			err = 0;
 			break;
 		default:
@@ -1034,16 +1039,12 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip,
 	}
 
 	/* create rawmidi device */
-	if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN) {
-		in_ports = out_ports = (int)quirk->data;
-	} else {
 	out_ports = 0;
 	in_ports = 0;
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables);
 		in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables);
 	}
-	}
 	err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports);
 	if (err < 0) {
 		snd_magic_kfree(umidi);
@@ -1052,7 +1053,7 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip,
 
 	/* create endpoint/port structures */
 	if (quirk && quirk->type == QUIRK_MIDI_MIDIMAN)
-		err = snd_usbmidi_create_endpoints_midiman(umidi, (int)quirk->data);
+		err = snd_usbmidi_create_endpoints_midiman(umidi, &endpoints[0]);
 	else
 		err = snd_usbmidi_create_endpoints(umidi, endpoints);
 	if (err < 0) {

sound/usb/usbmixer.c

@@ -81,8 +81,7 @@ struct usb_mixer_elem_info {
 	int channels;
 	int val_type;
 	int min, max, res;
-	unsigned int initialized: 1,
-		     hack_hole1: 1;		/* -256 value is missing */
+	unsigned int initialized: 1;
 };
 
 
@@ -281,17 +280,20 @@ static int get_ctl_value(usb_mixer_elem_info_t *cval, int request, int validx, i
 {
 	unsigned char buf[2];
 	int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1;
+	int timeout = 10;
  
+	while (timeout-- > 0) {
 		if (usb_control_msg(cval->chip->dev, usb_rcvctrlpipe(cval->chip->dev, 0),
 				    request,
 				    USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
 				    validx, cval->ctrlif | (cval->id << 8),
-			    buf, val_len, HZ) < 0) {
-		snd_printdd(KERN_ERR "cannot get ctl value: req = 0x%x, idx = 0x%x, val = 0x%x, type = %d\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type);
-		return -EINVAL;
-	}
+				    buf, val_len, HZ / 10) >= 0) {
 			*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
 			return 0;
+		}
+	}
+	snd_printdd(KERN_ERR "cannot get ctl value: req = 0x%x, idx = 0x%x, val = 0x%x, type = %d\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type);
+	return -EINVAL;
 }
 
 static int get_cur_ctl_value(usb_mixer_elem_info_t *cval, int validx, int *value)
@@ -313,15 +315,19 @@ static int set_ctl_value(usb_mixer_elem_info_t *cval, int request, int validx, i
 {
 	unsigned char buf[2];
 	int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1;
+	int timeout = 10;
  
 	value_set = convert_bytes_value(cval, value_set);
 	buf[0] = value_set & 0xff;
 	buf[1] = (value_set >> 8) & 0xff;
-	return usb_control_msg(cval->chip->dev, usb_sndctrlpipe(cval->chip->dev, 0),
+	while (timeout -- > 0)
+		if (usb_control_msg(cval->chip->dev, usb_sndctrlpipe(cval->chip->dev, 0),
 				    request,
 				    USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
 				    validx, cval->ctrlif | (cval->id << 8),
-			       buf, val_len, HZ);
+				    buf, val_len, HZ / 10) >= 0)
+			return 0;
+	return -EINVAL;
 }
 
 static int set_cur_ctl_value(usb_mixer_elem_info_t *cval, int validx, int value)

sound/usb/usbquirks.h

@@ -475,6 +475,20 @@
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x0033),
+	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
+		.vendor_name = "EDIROL",
+		.product_name = "PCR",
+		.ifnum = 0,
+		.type = QUIRK_MIDI_FIXED_ENDPOINT,
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.epnum = -1,
+			.out_cables = 0x0003,
+			.in_cables  = 0x0007
+		}
+	}
+},
 
 /* Midiman/M-Audio devices */
 {
@@ -484,7 +498,10 @@
 		.product_name = "MidiSport 2x2",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 2
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x0003,
+			.in_cables  = 0x0003
+		}
 	}
 },
 {
@@ -494,7 +511,10 @@
 		.product_name = "MidiSport 1x1",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 1
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x0001,
+			.in_cables  = 0x0001
+		}
 	}
 },
 {
@@ -504,7 +524,10 @@
 		.product_name = "Keystation",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 1
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x0001,
+			.in_cables  = 0x0001
+		}
 	}
 },
 {
@@ -514,7 +537,10 @@
 		.product_name = "MidiSport 4x4",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 4
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x000f,
+			.in_cables  = 0x000f
+		}
 	}
 },
 {
@@ -524,7 +550,23 @@
 		.product_name = "MidiSport 8x8",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 9
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x01ff,
+			.in_cables  = 0x01ff
+		}
+	}
+},
+{
+	USB_DEVICE_VENDOR_SPEC(0x0763, 0x1041),
+	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
+		.vendor_name = "M-Audio",
+		.product_name = "MidiSport 2x4",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_MIDI_MIDIMAN,
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x000f,
+			.in_cables  = 0x0003
+		}
 	}
 },
 {
@@ -534,7 +576,10 @@
 		.product_name = "Quattro",
 		.ifnum = 9,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 1
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x0001,
+			.in_cables  = 0x0001
+		}
 	}
 },
 {
@@ -544,7 +589,10 @@
 		.product_name = "AudioPhile",
 		.ifnum = 9,
 		.type = QUIRK_MIDI_MIDIMAN,
-		.data = (void*) 1
+		.data = & (const snd_usb_midi_endpoint_info_t) {
+			.out_cables = 0x0001,
+			.in_cables  = 0x0001
+		}
 	}
 },