ALSA 0.9.6 update

  - added __setup() to all midlevel modules
  - sequencer protocol 1.0.1
    - added timestamping flags for ports
  - OSS PCM emulation
    - fixed write() behaviour
    - added two new options no-silence & whole-frag
    - a try to fix OOPSes caused in the rate plugin
  - emu10k1 driver
    - more support for Audigy/Audigy2 EX
    - fixed soundfont locking
  - sb16 driver
    - fixed fm_res handling (and proc OOPS)
  - via82xx driver
    - fixed revision check for 8233A
  - usbaudio driver
    - added a workaround for M-Audio Audiophile USB

Documentation/sound/alsa/ALSA-Configuration.txt

@@ -1303,6 +1303,9 @@ Proc interfaces (/proc/asound)
 	  - direct    don't use plugins
 	  - block     force block mode (rvplayer)
 	  - non-block force non-block mode
+	  - whole-frag  write only whole fragments (optimization affecting
+			playback only)
+	  - no-silence  do not fill silence ahead to avoid clicks
 
   Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss
            echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss

include/sound/asequencer.h

@@ -29,7 +29,7 @@
 #include <sound/asound.h>
 
 /** version of the sequencer */
-#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 0)
+#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1)
 
 /**
  * definition of sequencer event types
@@ -604,6 +604,8 @@ struct sndrv_seq_remove_events {
 
 /* misc. conditioning flags */
 #define SNDRV_SEQ_PORT_FLG_GIVEN_PORT	(1<<0)
+#define SNDRV_SEQ_PORT_FLG_TIMESTAMP	(1<<1)
+#define SNDRV_SEQ_PORT_FLG_TIME_REAL	(1<<1)
 
 struct sndrv_seq_port_info {
 	struct sndrv_seq_addr addr;	/* client/port numbers */
@@ -620,7 +622,8 @@ struct sndrv_seq_port_info {
 
 	void *kernel;			/* reserved for kernel use (must be NULL) */
 	unsigned int flags;		/* misc. conditioning */
-	char reserved[60];		/* for future use */
+	unsigned char time_queue;	/* queue # for timestamping */
+	char reserved[59];		/* for future use */
 };
 
 

include/sound/asound.h

@@ -105,9 +105,10 @@ enum sndrv_hwdep_iface {
 	SNDRV_HWDEP_IFACE_ICS2115,	/* Wavetable synth */
 	SNDRV_HWDEP_IFACE_SSCAPE,	/* Ensoniq SoundScape ISA card (MC68EC000) */
 	SNDRV_HWDEP_IFACE_VX,		/* Digigram VX cards */
+	SNDRV_HWDEP_IFACE_MIXART,	/* Digigram miXart cards */
 
 	/* Don't forget to change the following: */
-	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_VX,
+	SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_MIXART,
 };
 
 struct sndrv_hwdep_info {

include/sound/emu10k1.h

@@ -931,6 +931,7 @@ struct _snd_emu10k1 {
 	unsigned long port;			/* I/O port number */
 	struct resource *res_port;
 	int APS: 1,				/* APS flag */
+	    no_ac97: 1,				/* no AC'97 */
 	    tos_link: 1;			/* tos link detected */
 	unsigned int audigy;			/* is Audigy? */
 	unsigned int revision;			/* chip revision */

include/sound/pcm_oss.h

@@ -30,7 +30,9 @@ struct _snd_pcm_oss_setup {
 	unsigned int disable:1,
 		     direct:1,
 		     block:1,
-		     nonblock:1;
+		     nonblock:1,
+		     wholefrag:1,
+		     nosilence:1;
 	unsigned int periods;
 	unsigned int period_size;
 	snd_pcm_oss_setup_t *next;

include/sound/soundfont.h

@@ -95,7 +95,6 @@ typedef struct snd_sf_list {
 	int zone_locked;	/* locked time for zone */
 	int sample_locked;	/* locked time for sample */
 	snd_sf_callback_t callback;	/* callback functions */
-	char sf_locked;		/* font lock flag */
 	struct semaphore presets_mutex;
 	spinlock_t lock;
 	snd_util_memhdr_t *memhdr;

include/sound/version.h

 /* include/version.h.  Generated by configure.  */
-#define CONFIG_SND_VERSION "0.9.5"
-#define CONFIG_SND_DATE " (Mon Jul 21 08:59:59 2003 UTC)"
+#define CONFIG_SND_VERSION "0.9.6"
+#define CONFIG_SND_DATE " (Mon Jul 28 11:08:42 2003 UTC)"

sound/core/memalloc.c

@@ -946,6 +946,25 @@ module_init(snd_mem_init)
 module_exit(snd_mem_exit)
 
 
+#ifndef MODULE
+
+/* format is: snd-page-alloc=enable */
+
+static int __init snd_mem_setup(char *str)
+{
+	static unsigned __initdata nr_dev = 0;
+
+	if (nr_dev >= SNDRV_CARDS)
+		return 0;
+	(void)(get_option(&str,&enable[nr_dev]) == 2);
+	nr_dev++;
+	return 1;
+}
+
+__setup("snd-page-alloc=", snd_mem_setup);
+
+#endif
+
 /*
  * exports
  */

sound/core/oss/pcm_oss.c

@@ -454,8 +454,18 @@ static int snd_pcm_oss_change_params(snd_pcm_substream_t *substream)
 	sw_params->sleep_min = 0;
 	sw_params->avail_min = runtime->period_size;
 	sw_params->xfer_align = 1;
-	sw_params->silence_threshold = 0;
-	sw_params->silence_size = 0;
+	if (atomic_read(&runtime->mmap_count) ||
+	    (substream->oss.setup && substream->oss.setup->nosilence)) {
+		sw_params->silence_threshold = 0;
+		sw_params->silence_size = 0;
+	} else {
+		snd_pcm_uframes_t frames;
+		frames = runtime->period_size + 16;
+		if (frames > runtime->buffer_size)
+			frames = runtime->buffer_size;
+		sw_params->silence_threshold = frames;
+		sw_params->silence_size = frames;
+	}
 
 	if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_SW_PARAMS, sw_params)) < 0) {
 		snd_printd("SW_PARAMS failed: %i\n", err);
@@ -581,7 +591,7 @@ static int snd_pcm_oss_capture_position_fixup(snd_pcm_substream_t *substream, sn
 		if (err < 0)
 			break;
 		runtime = substream->runtime;
-		if (*delay <= runtime->buffer_size)
+		if (*delay <= (snd_pcm_sframes_t)runtime->buffer_size)
 			break;
 		/* in case of overrun, skip whole periods like OSS/Linux driver does */
 		/* until avail(delay) <= buffer_size */
@@ -802,8 +812,9 @@ static ssize_t snd_pcm_oss_write1(snd_pcm_substream_t *substream, const char *bu
 			buf += tmp;
 			bytes -= tmp;
 			xfer += tmp;
-			if (runtime->oss.buffer_used == runtime->oss.period_bytes) {
-				tmp = snd_pcm_oss_write2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1);
+			if (substream->oss.setup == NULL || !substream->oss.setup->wholefrag ||
+			    runtime->oss.buffer_used == runtime->oss.period_bytes) {
+				tmp = snd_pcm_oss_write2(substream, runtime->oss.buffer, runtime->oss.buffer_used, 1);
 				if (tmp <= 0)
 					return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp;
 				runtime->oss.bytes += tmp;
@@ -2108,14 +2119,16 @@ static void snd_pcm_oss_proc_read(snd_info_entry_t *entry,
 	snd_pcm_oss_setup_t *setup = pstr->oss.setup_list;
 	down(&pstr->oss.setup_mutex);
 	while (setup) {
-		snd_iprintf(buffer, "%s %u %u%s%s%s%s\n",
+		snd_iprintf(buffer, "%s %u %u%s%s%s%s%s%s\n",
 			    setup->task_name,
 			    setup->periods,
 			    setup->period_size,
 			    setup->disable ? " disable" : "",
 			    setup->direct ? " direct" : "",
 			    setup->block ? " block" : "",
-			    setup->nonblock ? " non-block" : "");
+			    setup->nonblock ? " non-block" : "",
+			    setup->wholefrag ? " whole-frag" : "",
+			    setup->nosilence ? " no-silence" : "");
 		setup = setup->next;
 	}
 	up(&pstr->oss.setup_mutex);
@@ -2181,6 +2194,10 @@ static void snd_pcm_oss_proc_write(snd_info_entry_t *entry,
 				template.block = 1;
 			} else if (!strcmp(str, "non-block")) {
 				template.nonblock = 1;
+			} else if (!strcmp(str, "whole-frag")) {
+				template.wholefrag = 1;
+			} else if (!strcmp(str, "no-silence")) {
+				template.nosilence = 1;
 			}
 		} while (*str);
 		if (setup == NULL) {
@@ -2372,3 +2389,24 @@ static void __exit alsa_pcm_oss_exit(void)
 
 module_init(alsa_pcm_oss_init)
 module_exit(alsa_pcm_oss_exit)
+
+#ifndef MODULE
+
+/* format is: snd-pcm-oss=dsp_map,adsp_map[,nonblock_open] */
+
+static int __init alsa_pcm_oss_setup(char *str)
+{
+	static unsigned __initdata nr_dev = 0;
+
+	if (nr_dev >= SNDRV_CARDS)
+		return 0;
+	(void)(get_option(&str,&dsp_map[nr_dev]) == 2 &&
+	       get_option(&str,&adsp_map[nr_dev]) == 2);
+	(void)(get_option(&str,&nonblock_open) == 2);
+	nr_dev++;
+	return 1;
+}
+
+__setup("snd-pcm-oss=", alsa_pcm_oss_setup);
+
+#endif /* !MODULE */

sound/core/oss/pcm_plugin.c

@@ -56,6 +56,17 @@ static int snd_pcm_plugin_dst_channels_mask(snd_pcm_plugin_t *plugin,
 	return 0;
 }
 
+/*
+ *  because some cards might have rates "very close", we ignore
+ *  all "resampling" requests within +-5%
+ */
+static int rate_match(unsigned int src_rate, unsigned int dst_rate)
+{
+	unsigned int low = (src_rate * 95) / 100;
+	unsigned int high = (src_rate * 105) / 100;
+	return dst_rate >= low && dst_rate <= high;
+}
+
 static int snd_pcm_plugin_alloc(snd_pcm_plugin_t *plugin, snd_pcm_uframes_t frames)
 {
 	snd_pcm_plugin_format_t *format;
@@ -80,11 +91,14 @@ static int snd_pcm_plugin_alloc(snd_pcm_plugin_t *plugin, snd_pcm_uframes_t fram
 		plugin->buf = vmalloc(size);
 		plugin->buf_frames = frames;
 	}
-	if (!plugin->buf)
+	if (!plugin->buf) {
+		plugin->buf_frames = 0;
 		return -ENOMEM;
+	}
 	c = plugin->buf_channels;
 	if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) {
 		for (channel = 0; channel < format->channels; channel++, c++) {
+			c->frames = frames;
 			c->enabled = 1;
 			c->wanted = 0;
 			c->area.addr = plugin->buf;
@@ -95,6 +109,7 @@ static int snd_pcm_plugin_alloc(snd_pcm_plugin_t *plugin, snd_pcm_uframes_t fram
 		snd_assert((size % format->channels) == 0,);
 		size /= format->channels;
 		for (channel = 0; channel < format->channels; channel++, c++) {
+			c->frames = frames;
 			c->enabled = 1;
 			c->wanted = 0;
 			c->area.addr = plugin->buf + (channel * size);
@@ -420,7 +435,7 @@ int snd_pcm_plug_format_plugins(snd_pcm_plug_t *plug,
 
 	/* Format change (linearization) */
 	if ((srcformat.format != dstformat.format ||
-	     srcformat.rate != dstformat.rate ||
+	     !rate_match(srcformat.rate, dstformat.rate) ||
 	     srcformat.channels != dstformat.channels) &&
 	    !snd_pcm_format_linear(srcformat.format)) {
 		if (snd_pcm_format_linear(dstformat.format))
@@ -468,7 +483,7 @@ int snd_pcm_plug_format_plugins(snd_pcm_plug_t *plug,
 				ttable[v * sv + v] = FULL;
 		}
 		tmpformat.channels = dstformat.channels;
-		if (srcformat.rate == dstformat.rate &&
+		if (rate_match(srcformat.rate, dstformat.rate) &&
 		    snd_pcm_format_linear(dstformat.format))
 			tmpformat.format = dstformat.format;
 		err = snd_pcm_plugin_build_route(plug,
@@ -490,7 +505,7 @@ int snd_pcm_plug_format_plugins(snd_pcm_plug_t *plug,
 	}
 
 	/* rate resampling */
-	if (srcformat.rate != dstformat.rate) {
+	if (!rate_match(srcformat.rate, dstformat.rate)) {
 		tmpformat.rate = dstformat.rate;
 		if (srcformat.channels == dstformat.channels &&
 		    snd_pcm_format_linear(dstformat.format))

sound/core/oss/pcm_plugin.h

@@ -106,6 +106,7 @@ typedef struct _snd_pcm_channel_area {
 typedef struct _snd_pcm_plugin_channel {
 	void *aptr;			/* pointer to the allocated area */
 	snd_pcm_channel_area_t area;
+	snd_pcm_uframes_t frames;	/* allocated frames */
 	unsigned int enabled:1;		/* channel need to be processed */
 	unsigned int wanted:1;		/* channel is wanted */
 } snd_pcm_plugin_channel_t;

sound/core/oss/plugin_ops.h

@@ -323,7 +323,7 @@ get_s16_1234_xxC3: sample = swab16(as_u32(src) ^ 0x80); goto GET_S16_END;
 
 #ifdef PUT_S16_LABELS
 /* dst_wid dst_endswap unsigned */
-static void *put_s16_labels[4 * 2 * 2 * 4 * 2] = {
+static void *put_s16_labels[4 * 2 * 2] = {
 	&&put_s16_xx12_xxx1,	 /* 16h ->  8h */
 	&&put_s16_xx12_xxx9,	 /* 16h ^>  8h */
 	&&put_s16_xx12_xxx1,	 /* 16h ->  8s */

sound/core/oss/rate.c

@@ -85,11 +85,7 @@ static void resample_expand(snd_pcm_plugin_t *plugin,
 #undef PUT_S16_LABELS
 	void *get = get_s16_labels[data->get];
 	void *put = put_s16_labels[data->put];
-	void *get_s16_end = 0;
 	signed short sample = 0;
-#define GET_S16_END *get_s16_end
-#include "plugin_ops.h"
-#undef GET_S16_END
 	
 	for (channel = 0; channel < plugin->src_format.channels; channel++) {
 		pos = data->pos;
@@ -108,24 +104,16 @@ static void resample_expand(snd_pcm_plugin_t *plugin,
 		dst_step = dst_channels[channel].area.step / 8;
 		src_frames1 = src_frames;
 		dst_frames1 = dst_frames;
-		if (pos & ~R_MASK) {
-			get_s16_end = &&after_get1;
-			goto *get;
-		after_get1:
-			pos &= R_MASK;
-			S1 = S2;
-			S2 = sample;
-			src += src_step;
-			src_frames1--;
-		}
 		while (dst_frames1-- > 0) {
 			if (pos & ~R_MASK) {
 				pos &= R_MASK;
 				S1 = S2;
 				if (src_frames1-- > 0) {
-					get_s16_end = &&after_get2;
 					goto *get;
-				after_get2:
+#define GET_S16_END after_get
+#include "plugin_ops.h"
+#undef GET_S16_END
+				after_get:
 					S2 = sample;
 					src += src_step;
 				}
@@ -318,6 +306,8 @@ static snd_pcm_sframes_t rate_transfer(snd_pcm_plugin_t *plugin,
 #endif
 
 	dst_frames = rate_dst_frames(plugin, frames);
+	if (dst_frames > dst_channels[0].frames)
+		dst_frames = dst_channels[0].frames;
 	data = (rate_t *)plugin->extra_data;
 	data->func(plugin, src_channels, dst_channels, frames, dst_frames);
 	return dst_frames;

sound/core/oss/route.c

@@ -518,6 +518,10 @@ int getput_index(int format)
 	int sign, width, endian;
 	sign = !snd_pcm_format_signed(format);
 	width = snd_pcm_format_width(format) / 8 - 1;
+	if (width < 0 || width > 3) {
+		snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format);
+		width = 0;
+	}
 #ifdef SNDRV_LITTLE_ENDIAN
 	endian = snd_pcm_format_big_endian(format);
 #else

sound/core/pcm_lib.c

@@ -60,13 +60,11 @@ void snd_pcm_playback_silence(snd_pcm_substream_t *substream, snd_pcm_uframes_t
 			return;
 		snd_assert(runtime->silence_filled <= runtime->buffer_size, return);
 		noise_dist = snd_pcm_playback_hw_avail(runtime) + runtime->silence_filled;
-		if (noise_dist > (snd_pcm_sframes_t) runtime->silence_threshold)
+		if (noise_dist >= (snd_pcm_sframes_t) runtime->silence_threshold)
 			return;
 		frames = runtime->silence_threshold - noise_dist;
 		if (frames > runtime->silence_size)
 			frames = runtime->silence_size;
-		else
-			frames = runtime->silence_threshold;
 	} else {
 		if (new_hw_ptr == ULONG_MAX) {	/* initialization */
 			runtime->silence_filled = 0;
@@ -86,10 +84,9 @@ void snd_pcm_playback_silence(snd_pcm_substream_t *substream, snd_pcm_uframes_t
 			if ((snd_pcm_sframes_t)runtime->silence_start < 0)
 				runtime->silence_start += runtime->boundary;
 		}
-		frames = runtime->buffer_size;
+		frames = runtime->buffer_size - runtime->silence_filled;
 	}
-	snd_assert(frames >= runtime->silence_filled, return);
-	frames -= runtime->silence_filled;
+	snd_assert(frames <= runtime->buffer_size, return);
 	if (frames == 0)
 		return;
 	ofs = (runtime->silence_start + runtime->silence_filled) % runtime->buffer_size;

sound/core/pcm_memory.c

@@ -22,6 +22,7 @@
 #include <sound/driver.h>
 #include <asm/io.h>
 #include <linux/time.h>
+#include <linux/init.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/info.h>
@@ -568,3 +569,18 @@ struct page *snd_pcm_sgbuf_ops_page(snd_pcm_substream_t *substream, unsigned lon
 }
 
 #endif /* CONFIG_PCI */
+
+#ifndef MODULE
+
+/* format is: snd-pcm=preallocate_dma,maximum_substreams */
+
+static int __init alsa_pcm_setup(char *str)
+{
+	(void)(get_option(&str,&preallocate_dma) == 2 &&
+	       get_option(&str,&maximum_substreams) == 2);
+	return 1;
+}
+
+__setup("snd-pcm=", alsa_pcm_setup);
+
+#endif /* ifndef MODULE */

sound/core/pcm_native.c

@@ -461,9 +461,15 @@ static int snd_pcm_sw_params(snd_pcm_substream_t * substream, snd_pcm_sw_params_
 	if (params->xfer_align == 0 ||
 	    params->xfer_align % runtime->min_align != 0)
 		return -EINVAL;
-	if ((params->silence_threshold != 0 || params->silence_size < runtime->boundary) &&
-	    (params->silence_threshold + params->silence_size > runtime->buffer_size))
-		return -EINVAL;
+	if (params->silence_size >= runtime->boundary) {
+		if (params->silence_threshold != 0)
+			return -EINVAL;
+	} else {
+		if (params->silence_size > params->silence_threshold)
+			return -EINVAL;
+		if (params->silence_threshold > runtime->buffer_size)
+			return -EINVAL;
+	}
 	snd_pcm_stream_lock_irq(substream);
 	runtime->tstamp_mode = params->tstamp_mode;
 	runtime->sleep_min = params->sleep_min;

sound/core/rawmidi.c

@@ -1630,6 +1630,26 @@ static void __exit alsa_rawmidi_exit(void)
 module_init(alsa_rawmidi_init)
 module_exit(alsa_rawmidi_exit)
 
+#ifndef MODULE
+#ifdef CONFIG_SND_OSSEMUL
+/* format is: snd-rawmidi=midi_map,amidi_map */
+
+static int __init alsa_rawmidi_setup(char *str)
+{
+	static unsigned __initdata nr_dev = 0;
+
+	if (nr_dev >= SNDRV_CARDS)
+		return 0;
+	(void)(get_option(&str,&midi_map[nr_dev]) == 2 &&
+	       get_option(&str,&amidi_map[nr_dev]) == 2);
+	nr_dev++;
+	return 1;
+}
+
+__setup("snd-rawmidi=", alsa_rawmidi_setup);
+#endif /* CONFIG_SND_OSSEMUL */
+#endif /* ifndef MODULE */
+
 EXPORT_SYMBOL(snd_rawmidi_output_params);
 EXPORT_SYMBOL(snd_rawmidi_input_params);
 EXPORT_SYMBOL(snd_rawmidi_drop_output);

sound/core/rtctimer.c

@@ -57,7 +57,7 @@ static struct _snd_timer_hardware rtc_hw = {
 	.stop =		rtctimer_stop,
 };
 
-int rtctimer_freq = RTC_FREQ;		/* frequency */
+static int rtctimer_freq = RTC_FREQ;		/* frequency */
 static snd_timer_t *rtctimer;
 static atomic_t rtc_inc = ATOMIC_INIT(0);
 static rtc_task_t rtc_task;
@@ -182,4 +182,16 @@ MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz");
 
 MODULE_LICENSE("GPL");
 
+#ifndef MODULE
+/* format is: snd-rtctimer=freq */
+
+static int __init rtctimer_setup(char *str)
+{
+	(void)(get_option(&str,&rtctimer_freq) == 2);
+	return 1;
+}
+
+__setup("snd-rtctimer=", rtctimer_setup);
+#endif /* ifndef MODULE */
+
 #endif /* CONFIG_RTC || CONFIG_RTC_MODULE */

sound/core/seq/seq_clientmgr.c

@@ -520,6 +520,32 @@ static int bounce_error_event(client_t *client, snd_seq_event_t *event,
 }
 
 
+/*
+ * rewrite the time-stamp of the event record with the curren time
+ * of the given queue.
+ * return non-zero if updated.
+ */
+static int update_timestamp_of_queue(snd_seq_event_t *event, int queue, int real_time)
+{
+	queue_t *q;
+
+	q = queueptr(queue);
+	if (! q)
+		return 0;
+	event->queue = queue;
+	event->flags &= ~SNDRV_SEQ_TIME_STAMP_MASK;
+	if (real_time) {
+		event->time.time = snd_seq_timer_get_cur_time(q->timer);
+		event->flags |= SNDRV_SEQ_TIME_STAMP_REAL;
+	} else {
+		event->time.tick = snd_seq_timer_get_cur_tick(q->timer);
+		event->flags |= SNDRV_SEQ_TIME_STAMP_TICK;
+	}
+	queuefree(q);
+	return 1;
+}
+
+
 /*
  * deliver an event to the specified destination.
  * if filter is non-zero, client filter bitmap is tested.
@@ -551,6 +577,10 @@ static int snd_seq_deliver_single_event(client_t *client,
 		goto __skip;
 	}
 		
+	if (dest_port->timestamping)
+		update_timestamp_of_queue(event, dest_port->time_queue,
+					  dest_port->time_real);
+
 	/* expand the quoted event */
 	if (event->type == SNDRV_SEQ_EVENT_KERNEL_QUOTE) {
 		quoted = 1;
@@ -597,27 +627,6 @@ static int snd_seq_deliver_single_event(client_t *client,
 }
 
 
-static void snd_seq_subs_update_event_header(subscribers_t *subs, snd_seq_event_t *event)
-{
-	if (subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP) {
-		/* convert time according to flag with subscription */
-		queue_t *q;
-		q = queueptr(subs->info.queue);
-		if (q) {
-			event->queue = subs->info.queue;
-			event->flags &= ~SNDRV_SEQ_TIME_STAMP_MASK;
-			if (subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL) {
-				event->time.time = snd_seq_timer_get_cur_time(q->timer);
-				event->flags |= SNDRV_SEQ_TIME_STAMP_REAL;
-			} else {
-				event->time.tick = snd_seq_timer_get_cur_tick(q->timer);
-				event->flags |= SNDRV_SEQ_TIME_STAMP_TICK;
-			}
-			queuefree(q);
-		}
-	}
-}
-
 /*
  * send the event to all subscribers:
  */
@@ -647,7 +656,10 @@ static int deliver_to_subscribers(client_t *client,
 	list_for_each(p, &grp->list_head) {
 		subs = list_entry(p, subscribers_t, src_list);
 		event->dest = subs->info.dest;
-		snd_seq_subs_update_event_header(subs, event);
+		if (subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP)
+			/* convert time according to flag with subscription */
+			update_timestamp_of_queue(event, subs->info.queue,
+						  subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL);
 		err = snd_seq_deliver_single_event(client, event,
 						   0, atomic, hop);
 		if (err < 0)

sound/core/seq/seq_ports.c

@@ -352,6 +352,11 @@ int snd_seq_set_port_info(client_port_t * port, snd_seq_port_info_t * info)
 	port->midi_voices = info->midi_voices;
 	port->synth_voices = info->synth_voices;
 
+	/* timestamping */
+	port->timestamping = (info->flags & SNDRV_SEQ_PORT_FLG_TIMESTAMP) ? 1 : 0;
+	port->time_real = (info->flags & SNDRV_SEQ_PORT_FLG_TIME_REAL) ? 1 : 0;
+	port->time_queue = info->time_queue;
+
 	return 0;
 }
 
@@ -378,6 +383,15 @@ int snd_seq_get_port_info(client_port_t * port, snd_seq_port_info_t * info)
 	info->read_use = port->c_src.count;
 	info->write_use = port->c_dest.count;
 	
+	/* timestamping */
+	info->flags = 0;
+	if (port->timestamping) {
+		info->flags |= SNDRV_SEQ_PORT_FLG_TIMESTAMP;
+		if (port->time_real)
+			info->flags |= SNDRV_SEQ_PORT_FLG_TIME_REAL;
+		info->time_queue = port->time_queue;
+	}
+
 	return 0;
 }
 

sound/core/seq/seq_ports.h

@@ -74,6 +74,9 @@ typedef struct client_port_t {
 	void *private_data;
 	unsigned int callback_all : 1;
 	unsigned int closing : 1;
+	unsigned int timestamping: 1;
+	unsigned int time_real: 1;
+	int time_queue;
 	
 	/* capability, inport, output, sync */
 	unsigned int capability;	/* port capability bits */

sound/core/sound.c

@@ -389,6 +389,24 @@ static void __exit alsa_sound_exit(void)
 module_init(alsa_sound_init)
 module_exit(alsa_sound_exit)
 
+#ifndef MODULE
+
+/* format is: snd=major,cards_limit[,device_mode] */
+
+static int __init alsa_sound_setup(char *str)
+{
+	(void)(get_option(&str,&major) == 2 &&
+	       get_option(&str,&cards_limit) == 2);
+#ifdef CONFIG_DEVFS_FS
+	(void)(get_option(&str,&device_mode) == 2);
+#endif
+	return 1;
+}
+
+__setup("snd=", alsa_sound_setup);
+
+#endif /* ifndef MODULE */
+
   /* sound.c */
 EXPORT_SYMBOL(snd_major);
 EXPORT_SYMBOL(snd_ecards_limit);

sound/core/timer.c

@@ -41,7 +41,7 @@
 #define DEFAULT_TIMER_LIMIT 2
 #endif
 
-int timer_limit = DEFAULT_TIMER_LIMIT;
+static int timer_limit = DEFAULT_TIMER_LIMIT;
 MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("ALSA timer interface");
 MODULE_LICENSE("GPL");
@@ -1806,6 +1806,18 @@ static void __exit alsa_timer_exit(void)
 module_init(alsa_timer_init)
 module_exit(alsa_timer_exit)
 
+#ifndef MODULE
+/* format is: snd-timer=timer_limit */
+
+static int __init alsa_timer_setup(char *str)
+{
+	(void)(get_option(&str,&timer_limit) == 2);
+	return 1;
+}
+
+__setup("snd-timer=", alsa_timer_setup);
+#endif /* ifndef MODULE */
+
 EXPORT_SYMBOL(snd_timer_open);
 EXPORT_SYMBOL(snd_timer_close);
 EXPORT_SYMBOL(snd_timer_resolution);

sound/drivers/dummy.c

@@ -69,6 +69,15 @@ MODULE_DEVICES("{{ALSA,Dummy soundcard}}");
 #define USE_PERIODS_MAX		255
 #endif
 
+#if 0 /* simple AC97 bridge (intel8x0) with 48kHz AC97 only codec */
+#define USE_FORMATS		SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN	2
+#define USE_CHANNELS_MAX	2
+#define USE_RATE		SNDRV_PCM_RATE_48000
+#define USE_RATE_MIN		48000
+#define USE_RATE_MAX		48000
+#endif
+
 
 /* defaults */
 #ifndef MAX_BUFFER_SIZE
@@ -77,6 +86,11 @@ MODULE_DEVICES("{{ALSA,Dummy soundcard}}");
 #ifndef USE_FORMATS
 #define USE_FORMATS 		(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
 #endif
+#ifndef USE_RATE
+#define USE_RATE		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000
+#define USE_RATE_MIN		5500
+#define USE_RATE_MAX		48000
+#endif
 #ifndef USE_CHANNELS_MIN
 #define USE_CHANNELS_MIN 	1
 #endif
@@ -271,9 +285,9 @@ static snd_pcm_hardware_t snd_card_dummy_playback =
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID),
 	.formats =		USE_FORMATS,
-	.rates =		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
-	.rate_min =		5500,
-	.rate_max =		48000,
+	.rates =		USE_RATE,
+	.rate_min =		USE_RATE_MIN,
+	.rate_max =		USE_RATE_MAX,
 	.channels_min =		USE_CHANNELS_MIN,
 	.channels_max =		USE_CHANNELS_MAX,
 	.buffer_bytes_max =	MAX_BUFFER_SIZE,
@@ -289,9 +303,9 @@ static snd_pcm_hardware_t snd_card_dummy_capture =
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID),
 	.formats =		USE_FORMATS,
-	.rates =		SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000,
-	.rate_min =		5500,
-	.rate_max =		48000,
+	.rates =		USE_RATE,
+	.rate_min =		USE_RATE_MIN,
+	.rate_max =		USE_RATE_MAX,
 	.channels_min =		USE_CHANNELS_MIN,
 	.channels_max =		USE_CHANNELS_MAX,
 	.buffer_bytes_max =	MAX_BUFFER_SIZE,

sound/isa/sb/sb16.c

@@ -350,6 +350,18 @@ static int __devinit snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard,
 
 #endif /* CONFIG_PNP */
 
+static void snd_sb16_free(snd_card_t *card)
+{
+	struct snd_card_sb16 *acard = (struct snd_card_sb16 *)card->private_data;
+        
+	if (acard == NULL)
+		return;
+	if (acard->fm_res) {
+		release_resource(acard->fm_res);
+		kfree_nocheck(acard->fm_res);
+	}
+}
+
 static int __init snd_sb16_probe(int dev,
 				 struct pnp_card_link *pcard,
 				 const struct pnp_card_device_id *pid)
@@ -374,6 +386,7 @@ static int __init snd_sb16_probe(int dev,
 	if (card == NULL)
 		return -ENOMEM;
 	acard = (struct snd_card_sb16 *) card->private_data;
+	card->private_free = snd_sb16_free;
 #ifdef CONFIG_PNP
 	if (isapnp[dev]) {
 		if ((err = snd_card_sb16_pnp(dev, acard, pcard, pid))) {
@@ -464,7 +477,8 @@ static int __init snd_sb16_probe(int dev,
 
 	if (fm_port[dev] > 0) {
 		if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2,
-				    OPL3_HW_OPL3, fm_port[dev] == port[dev],
+				    OPL3_HW_OPL3,
+				    fm_port[dev] == port[dev] || fm_port[dev] == 0x388,
 				    &opl3) < 0) {
 			snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n",
 				   fm_port[dev], fm_port[dev] + 2);

sound/pci/ac97/ac97_patch.c

@@ -650,6 +650,26 @@ int patch_ad1881(ac97_t * ac97)
 	return 0;
 }
 
+static const snd_kcontrol_new_t snd_ac97_controls_ad1885[] = {
+	AC97_SINGLE("Digital Mono Direct", AC97_AD_MISC, 11, 1, 0),
+	AC97_SINGLE("Digital Audio Mode", AC97_AD_MISC, 12, 1, 0),
+	AC97_SINGLE("Low Power Mixer", AC97_AD_MISC, 14, 1, 0),
+	AC97_SINGLE("Zero Fill DAC", AC97_AD_MISC, 15, 1, 0),
+};
+
+static int patch_ad1885_specific(ac97_t * ac97)
+{
+	int err;
+
+	if ((err = patch_build_controls(ac97, snd_ac97_controls_ad1885, ARRAY_SIZE(snd_ac97_controls_ad1885))) < 0)
+		return err;
+	return 0;
+}
+
+static struct snd_ac97_build_ops patch_ad1885_build_ops = {
+	.build_specific = &patch_ad1885_specific
+};
+
 int patch_ad1885(ac97_t * ac97)
 {
 	unsigned short jack;
@@ -661,6 +681,8 @@ int patch_ad1885(ac97_t * ac97)
 	/* turn off jack sense bits D8 & D9 */
 	jack = snd_ac97_read(ac97, AC97_AD_JACK_SPDIF);
 	snd_ac97_write_cache(ac97, AC97_AD_JACK_SPDIF, jack | 0x0300);
+
+	ac97->build_ops = &patch_ad1885_build_ops;
 	return 0;
 }
 

sound/pci/emu10k1/emu10k1.c

@@ -135,11 +135,9 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		snd_card_free(card);
 		return err;
 	}		
-	if (!emu->APS) {	/* APS board has not an AC97 mixer */
-		if ((err = snd_emu10k1_mixer(emu)) < 0) {
-			snd_card_free(card);
-			return err;
-		}		
+	if ((err = snd_emu10k1_mixer(emu)) < 0) {
+		snd_card_free(card);
+		return err;
 	}
 	if (emu->audigy) {
 		if ((err = snd_emu10k1_audigy_midi(emu)) < 0) {

sound/pci/emu10k1/emu10k1_main.c

@@ -674,6 +674,14 @@ int __devinit snd_emu10k1_create(snd_card_t * card,
 	if (emu->serial == 0x40011102) {
 		emu->card_type = EMU10K1_CARD_EMUAPS;
 		emu->APS = 1;
+		emu->no_ac97 = 1; /* APS has no AC97 chip */
+	}
+	else if (emu->revision == 4 && emu->serial == 0x10051102) {
+		/* Audigy 2 EX has apparently no effective AC97 controls
+		 * (for both input and output), so we skip the AC97 detections
+		 */
+		snd_printdd(KERN_INFO "Audigy2 EX is detected. skpping ac97.\n");
+		emu->no_ac97 = 1;
 	}
 	
 	emu->fx8010.fxbus_mask = 0x303f;

sound/pci/emu10k1/emufx.c

@@ -1238,7 +1238,10 @@ static void __devinit snd_emu10k1_init_stereo_onoff_control(emu10k1_fx8010_contr
 
 static int __devinit _snd_emu10k1_audigy_init_efx(emu10k1_t *emu)
 {
-	int err, i, z, gpr, tmp, playback, capture, nctl;
+	int err, i, z, gpr, nctl;
+	const int playback = 10;
+	const int capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2); /* we reserve 10 voices */
+	const int tmp = 0x88;
 	u32 ptr;
 	emu10k1_fx8010_code_t *icode;
 	emu10k1_fx8010_control_gpr_t *controls, *ctl;
@@ -1261,19 +1264,15 @@ static int __devinit _snd_emu10k1_audigy_init_efx(emu10k1_t *emu)
 	strcpy(icode->name, "Audigy DSP code for ALSA");
 	ptr = 0;
 	nctl = 0;
-	playback = 10;
-	capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2); /* we reserve 10 voices */
 	gpr = capture + 10;
-	tmp = 0x88;
 
 	/* stop FX processor */
 	snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP);
 
-	/* Wave Playback */
+	/* Wave Playback Volume */
 	A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT));
 	A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT));
-	snd_emu10k1_init_stereo_control(&controls[nctl++], "Wave Playback Volume", gpr,
-					emu->revision == 4 ? 50 : 100);
+	snd_emu10k1_init_stereo_control(&controls[nctl++], "Wave Playback Volume", gpr, 100);
 	gpr += 2;
 
 	/* Wave Surround Playback */
@@ -1497,6 +1496,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 	snd_emu10k1_init_stereo_onoff_control(controls + nctl++, "Tone Control - Switch", gpr, 0);
 	gpr += 2;
 
+	/* Master volume for audigy2 */
+	if (emu->revision == 4) {
+		A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS));
+		A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr+1), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS));
+		snd_emu10k1_init_stereo_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0);
+		gpr += 2;
+	}
+
 	/* digital outputs */
 	A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);
 	A_PUT_STEREO_OUTPUT(A_EXTOUT_REAR_L, A_EXTOUT_REAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS);
@@ -1504,7 +1511,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 	A_PUT_OUTPUT(A_EXTOUT_LFE, playback+5 + SND_EMU10K1_PLAYBACK_CHANNELS);
 
 	/* analog speakers */
-	if (emu->audigy && emu->revision == 4) { /* audigy2 */
+	if (emu->revision == 4) { /* audigy2 */
 		A_PUT_STEREO_OUTPUT(A_EXTOUT_AFRONT_L, A_EXTOUT_AFRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);
 	} else {
 		A_PUT_STEREO_OUTPUT(A_EXTOUT_AC97_L, A_EXTOUT_AC97_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS);

sound/pci/emu10k1/emumixer.c

@@ -423,6 +423,36 @@ static void snd_emu10k1_mixer_free_ac97(ac97_t *ac97)
 	emu->ac97 = NULL;
 }
 
+/*
+ */
+static int remove_ctl(snd_card_t *card, const char *name)
+{
+	snd_ctl_elem_id_t id;
+	memset(&id, 0, sizeof(id));
+	strcpy(id.name, name);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_remove_id(card, &id);
+}
+
+static snd_kcontrol_t *ctl_find(snd_card_t *card, const char *name)
+{
+	snd_ctl_elem_id_t sid;
+	memset(&sid, 0, sizeof(sid));
+	strcpy(sid.name, name);
+	sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_find_id(card, &sid);
+}
+
+static int rename_ctl(snd_card_t *card, const char *src, const char *dst)
+{
+	snd_kcontrol_t *kctl = ctl_find(card, src);
+	if (kctl) {
+		strcpy(kctl->id.name, dst);
+		return 0;
+	}
+	return -ENOENT;
+}
+
 int __devinit snd_emu10k1_mixer(emu10k1_t *emu)
 {
 	ac97_t ac97;
@@ -430,7 +460,7 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu)
 	snd_kcontrol_t *kctl;
 	snd_card_t *card = emu->card;
 
-	if (!emu->APS) {
+	if (!emu->no_ac97) {
 		memset(&ac97, 0, sizeof(ac97));
 		ac97.write = snd_emu10k1_ac97_write;
 		ac97.read = snd_emu10k1_ac97_read;
@@ -438,8 +468,33 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu)
 		ac97.private_free = snd_emu10k1_mixer_free_ac97;
 		if ((err = snd_ac97_mixer(emu->card, &ac97, &emu->ac97)) < 0)
 			return err;
+		if (emu->audigy && emu->revision == 4) {
+			/* Master/PCM controls on ac97 of Audigy2 has no effect */
+			/* FIXME: keep master volume/switch to be sure.
+			 * once after we check that they play really no roles,
+			 * they shall be removed.
+			 */
+			rename_ctl(card, "Master Playback Switch", "AC97 Master Playback Switch");
+			rename_ctl(card, "Master Playback Volume", "AC97 Master Playback Volume");
+			/* pcm controls are removed */
+			remove_ctl(card, "PCM Playback Switch");
+			remove_ctl(card, "PCM Playback Volume");
+		}
 	} else {
-		strcpy(emu->card->mixername, "EMU APS");
+		if (emu->APS)
+			strcpy(emu->card->mixername, "EMU APS");
+		else if (emu->audigy)
+			strcpy(emu->card->mixername, "SB Audigy");
+		else
+			strcpy(emu->card->mixername, "Emu10k1");
+	}
+
+	if (emu->audigy && emu->revision == 4) {
+		/* Audigy2 and Audigy2 EX */
+		/* use the conventional names */
+		rename_ctl(card, "Wave Playback Volume", "PCM Playback Volume");
+		rename_ctl(card, "Wave Playback Volume", "PCM Capture Volume");
+		rename_ctl(card, "Wave Master Playback Volume", "Master Playback Volume");
 	}
 
 	if ((kctl = emu->ctl_send_routing = snd_ctl_new1(&snd_emu10k1_send_routing_control, emu)) == NULL)
@@ -455,6 +510,7 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu)
 	if ((err = snd_ctl_add(card, kctl)))
 		return err;
 
+	/* intiailize the routing and volume table for each pcm playback stream */
 	for (pcm = 0; pcm < 32; pcm++) {
 		emu10k1_pcm_mixer_t *mix;
 		int v;
@@ -474,21 +530,25 @@ int __devinit snd_emu10k1_mixer(emu10k1_t *emu)
 		mix->attn[0] = mix->attn[1] = mix->attn[2] = 0xffff;
 	}
 	
-	if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL)
-		return -ENOMEM;
-	if ((err = snd_ctl_add(card, kctl)))
-		return err;
-	if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_control, emu)) == NULL)
-		return -ENOMEM;
-	if ((err = snd_ctl_add(card, kctl)))
-		return err;
+	if (! emu->APS) { /* FIXME: APS has these controls? */
+		/* sb live! and audigy */
+		if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL)
+			return -ENOMEM;
+		if ((err = snd_ctl_add(card, kctl)))
+			return err;
+		if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_control, emu)) == NULL)
+			return -ENOMEM;
+		if ((err = snd_ctl_add(card, kctl)))
+			return err;
+	}
 
 	if (emu->audigy) {
 		if ((kctl = snd_ctl_new1(&snd_audigy_shared_spdif, emu)) == NULL)
 			return -ENOMEM;
 		if ((err = snd_ctl_add(card, kctl)))
 			return err;
-	} else {
+	} else if (! emu->APS) {
+		/* sb live! */
 		if ((kctl = snd_ctl_new1(&snd_emu10k1_shared_spdif, emu)) == NULL)
 			return -ENOMEM;
 		if ((err = snd_ctl_add(card, kctl)))

sound/pci/emu10k1/irq.c

@@ -55,15 +55,13 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id, struct pt_regs *regs)
 		if (status & IPR_CHANNELLOOP) {
 			int voice;
 			int voice_max = status & IPR_CHANNELNUMBERMASK;
-			int voice_max_l;
 			u32 val;
 			emu10k1_voice_t *pvoice = emu->voices;
 
 			val = snd_emu10k1_ptr_read(emu, CLIPL, 0);
-			voice_max_l = voice_max;
-			if (voice_max_l >= 0x20)
-				voice_max_l = 0x1f;
-			for (voice = 0; voice <= voice_max_l; voice++) {
+			for (voice = 0; voice <= voice_max; voice++) {
+				if (voice == 0x20)
+					val = snd_emu10k1_ptr_read(emu, CLIPH, 0);
 				if (val & 1) {
 					if (pvoice->use && pvoice->interrupt != NULL) {
 						pvoice->interrupt(emu, pvoice);
@@ -75,21 +73,6 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id, struct pt_regs *regs)
 				val >>= 1;
 				pvoice++;
 			}
-			if (voice_max > 0x1f) {
-				val = snd_emu10k1_ptr_read(emu, CLIPH, 0);
-				for (; voice <= voice_max; voice++) {
-					if(val & 1) {
-						if (pvoice->use && pvoice->interrupt != NULL) {
-							pvoice->interrupt(emu, pvoice);
-							snd_emu10k1_voice_intr_ack(emu, voice);
-						} else {
-							snd_emu10k1_voice_intr_disable(emu, voice);
-						}
-					}
-					val >>= 1;
-					pvoice++;
-				}
-			}
 			status &= ~IPR_CHANNELLOOP;
 		}
 		status &= ~IPR_CHANNELNUMBERMASK;
@@ -150,9 +133,27 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id, struct pt_regs *regs)
 			status &= ~IPR_FXDSP;
 		}
 		if (status) {
-			snd_printd(KERN_WARNING "emu10k1: unhandled interrupt: 0x%08x\n", status);
+			unsigned int bits;
+			snd_printk(KERN_ERR "emu10k1: unhandled interrupt: 0x%08x\n", status);
+			//make sure any interrupts we don't handle are disabled:
+			bits = INTE_FXDSPENABLE |
+				INTE_PCIERRORENABLE |
+				INTE_VOLINCRENABLE |
+				INTE_VOLDECRENABLE |
+				INTE_MUTEENABLE |
+				INTE_MICBUFENABLE |
+				INTE_ADCBUFENABLE |
+				INTE_EFXBUFENABLE |
+				INTE_GPSPDIFENABLE |
+				INTE_CDSPDIFENABLE |
+				INTE_INTERVALTIMERENB |
+				INTE_MIDITXENABLE |
+				INTE_MIDIRXENABLE;
+			if (emu->audigy)
+				bits |= INTE_A_MIDITXENABLE2 | INTE_A_MIDIRXENABLE2;
+			snd_emu10k1_intr_disable(emu, bits);
 		}
-		outl(orig_status, emu->port + IPR); /* ack */
+		outl(orig_status, emu->port + IPR); /* ack all */
 	}
 	return IRQ_RETVAL(handled);
 }

sound/pci/intel8x0.c

@@ -1002,7 +1002,7 @@ static int snd_intel8x0_pcm_open(snd_pcm_substream_t * substream, ichdev_t *ichd
 	snd_pcm_runtime_t *runtime = substream->runtime;
 	static unsigned int i, rates[] = {
 		/* ATTENTION: these values depend on the definition in pcm.h! */
-		5512, 8000, 11025, 16000, 22050, 32000, 44100, 480000
+		5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000
 	};
 	int err;
 
@@ -1558,6 +1558,7 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	{ 0x1028, 0x0126, "Dell Optiplex GX260", AC97_TUNE_HP_ONLY },
 	{ 0x1734, 0x0088, "Fujitsu-Siemens D1522", AC97_TUNE_HP_ONLY },
 	{ 0x10f1, 0x2665, "Fujitsu-Siemens Celcius", AC97_TUNE_HP_ONLY },
+	{ 0x110a, 0x0056, "Fujitsu-Siemens Scenic", AC97_TUNE_HP_ONLY },
 	{ 0x8086, 0x4d44, "Intel D850EMV2", AC97_TUNE_HP_ONLY },
 	/* { 0x4144, 0x5360, "AMD64 Motherboard", AC97_TUNE_HP_ONLY }, */ /* FIXME: this seems invalid */
 	{ 0x1043, 0x80b0, "ASUS P4PE Mobo", AC97_TUNE_SWAP_SURROUND },

sound/pci/maestro3.c

@@ -1535,9 +1535,14 @@ static snd_pcm_uframes_t
 snd_m3_pcm_pointer(snd_pcm_substream_t * subs)
 {
 	m3_t *chip = snd_pcm_substream_chip(subs);
+	unsigned int ptr;
 	m3_dma_t *s = (m3_dma_t*)subs->runtime->private_data;
 	snd_assert(s != NULL, return 0);
-	return bytes_to_frames(subs->runtime, snd_m3_get_pointer(chip, s, subs));
+
+	spin_lock(&chip->reg_lock);
+	ptr = snd_m3_get_pointer(chip, s, subs);
+	spin_unlock(&chip->reg_lock);
+	return bytes_to_frames(subs->runtime, ptr);
 }
 
 

sound/pci/via82xx.c

@@ -886,7 +886,7 @@ static int snd_via8233_playback_prepare(snd_pcm_substream_t *substream)
 		snd_ac97_set_rate(chip->ac97, AC97_PCM_LFE_DAC_RATE, runtime->rate);
 		snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate);
 	}
-	if (chip->chip_type == TYPE_VIA8233A)
+	if (chip->revision == VIA_REV_8233A)
 		rbits = 0;
 	else
 		rbits = (0xfffff / 48000) * runtime->rate + ((0xfffff % 48000) * runtime->rate) / 48000;
@@ -928,7 +928,7 @@ static int snd_via8233_multi_prepare(snd_pcm_substream_t *substream)
 	fmt = (runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? VIA_REG_MULTPLAY_FMT_16BIT : VIA_REG_MULTPLAY_FMT_8BIT;
 	fmt |= runtime->channels << 4;
 	outb(fmt, VIADEV_REG(viadev, OFS_MULTPLAY_FORMAT));
-	if (chip->chip_type == TYPE_VIA8233A)
+	if (chip->revision == VIA_REV_8233A)
 		slots = 0;
 	else {
 		/* set sample number to slot 3, 4, 7, 8, 6, 9 (for VIA8233/C,8235) */
@@ -1108,7 +1108,7 @@ static int snd_via8233_multi_open(snd_pcm_substream_t * substream)
 	if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0)
 		return err;
 	substream->runtime->hw.channels_max = 6;
-	if (chip->chip_type == TYPE_VIA8233A)
+	if (chip->revision == VIA_REV_8233A)
 		snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &hw_constraints_channels);
 	return 0;
 }

sound/synth/emux/soundfont.c

@@ -66,15 +66,11 @@ static void snd_sf_clear(snd_sf_list_t *sflist);
 static int
 lock_preset(snd_sf_list_t *sflist, int nonblock)
 {
-	unsigned long flags;
-	spin_lock_irqsave(&sflist->lock, flags);
-	if (sflist->sf_locked && nonblock) {
-		spin_unlock_irqrestore(&sflist->lock, flags);
-		return -EBUSY;
-	}
-	spin_unlock_irqrestore(&sflist->lock, flags);
-	down(&sflist->presets_mutex);
-	sflist->sf_locked = 1;
+	if (nonblock) {
+		if (down_trylock(&sflist->presets_mutex))
+			return -EBUSY;
+	} else 
+		down(&sflist->presets_mutex);
 	return 0;
 }
 
@@ -86,7 +82,6 @@ static void
 unlock_preset(snd_sf_list_t *sflist)
 {
 	up(&sflist->presets_mutex);
-	sflist->sf_locked = 0;
 }
 
 
@@ -1356,7 +1351,6 @@ snd_sf_new(snd_sf_callback_t *callback, snd_util_memhdr_t *hdr)
 
 	init_MUTEX(&sflist->presets_mutex);
 	spin_lock_init(&sflist->lock);
-	sflist->sf_locked = 0;
 	sflist->memhdr = hdr;
 
 	if (callback)
@@ -1403,7 +1397,7 @@ snd_soundfont_remove_samples(snd_sf_list_t *sflist)
 
 /*
  * Remove unlocked samples.
- * The soundcard should be silet before calling this function.
+ * The soundcard should be silent before calling this function.
  */
 int
 snd_soundfont_remove_unlocked(snd_sf_list_t *sflist)

sound/usb/usbaudio.c

@@ -2297,6 +2297,14 @@ static int parse_audio_endpoints(snd_usb_audio_t *chip, int iface_no)
 			 */
 			fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE;
 		}
+
+		/* workaround for M-Audio Audiophile USB */
+		if (dev->descriptor.idVendor == 0x0763 &&
+		    dev->descriptor.idProduct == 0x2003) {
+			/* doesn't set the sample rate attribute, but supports it */
+			fp->attributes |= EP_CS_ATTR_SAMPLE_RATE;
+		}
+
 		/*
 		 * plantronics headset and Griffin iMic have set adaptive-in
 		 * although it's really not...