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-rw-r--r--sound/arm/aaci.c2
-rw-r--r--sound/core/pcm_lib.c10
-rw-r--r--sound/core/pcm_native.c6
-rw-r--r--sound/drivers/pcsp/pcsp_mixer.c2
-rw-r--r--sound/oss/Kconfig2
-rw-r--r--sound/oss/sh_dac_audio.c85
-rw-r--r--sound/pci/ac97/ac97_patch.c7
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c2
-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/pci/hda/patch_realtek.c7
-rw-r--r--sound/pci/hda/patch_sigmatel.c10
-rw-r--r--sound/usb/usbaudio.c2
-rw-r--r--sound/usb/usbaudio.h2
-rw-r--r--sound/usb/usbmidi.c12
-rw-r--r--sound/usb/usbquirks.h2
16 files changed, 88 insertions, 65 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7fbd68fab94..5c48e36038f 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1074,7 +1074,7 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
return i;
}
-static int __devinit aaci_probe(struct amba_device *dev, void *id)
+static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id)
{
struct aaci *aaci;
int ret, i;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a2a792c18c4..d659995ac3a 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -249,6 +249,11 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
new_hw_ptr = hw_base + pos;
}
}
+
+ /* Do jiffies check only in xrun_debug mode */
+ if (!xrun_debug(substream))
+ goto no_jiffies_check;
+
/* Skip the jiffies check for hardwares with BATCH flag.
* Such hardware usually just increases the position at each IRQ,
* thus it can't give any strange position.
@@ -336,7 +341,9 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
hw_base = 0;
new_hw_ptr = hw_base + pos;
}
- if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
+ /* Do jiffies check only in xrun_debug mode */
+ if (xrun_debug(substream) &&
+ ((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
hw_ptr_error(substream,
"hw_ptr skipping! "
"(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n",
@@ -1478,7 +1485,6 @@ static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream,
runtime->status->hw_ptr %= runtime->buffer_size;
else
runtime->status->hw_ptr = 0;
- runtime->hw_ptr_jiffies = jiffies;
snd_pcm_stream_unlock_irqrestore(substream, flags);
return 0;
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index fc6f98e257d..b5da656d1ec 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -848,6 +848,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
{
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_trigger_tstamp(substream);
+ runtime->hw_ptr_jiffies = jiffies;
runtime->status->state = state;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
@@ -961,6 +962,11 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
{
if (substream->runtime->trigger_master != substream)
return 0;
+ /* The jiffies check in snd_pcm_update_hw_ptr*() is done by
+ * a delta betwen the current jiffies, this gives a large enough
+ * delta, effectively to skip the check once.
+ */
+ substream->runtime->hw_ptr_jiffies = jiffies - HZ * 1000;
return substream->ops->trigger(substream,
push ? SNDRV_PCM_TRIGGER_PAUSE_PUSH :
SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 771955a9be7..199b0337714 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -51,7 +51,7 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
if (uinfo->value.enumerated.item > chip->max_treble)
uinfo->value.enumerated.item = chip->max_treble;
sprintf(uinfo->value.enumerated.name, "%lu",
- PCSP_CALC_RATE(uinfo->value.enumerated.item));
+ (unsigned long)PCSP_CALC_RATE(uinfo->value.enumerated.item));
return 0;
}
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 1ca7427c4b6..bcf2a0698d5 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -561,7 +561,7 @@ endif # SOUND_OSS
config SOUND_SH_DAC_AUDIO
tristate "SuperH DAC audio support"
- depends on CPU_SH3
+ depends on CPU_SH3 && HIGH_RES_TIMERS
config SOUND_SH_DAC_AUDIO_CHANNEL
int "DAC channel"
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index 78cfb66e4c5..b2ed8757542 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -18,47 +18,36 @@
#include <linux/sound.h>
#include <linux/soundcard.h>
#include <linux/interrupt.h>
+#include <linux/hrtimer.h>
#include <asm/io.h>
#include <asm/uaccess.h>
#include <asm/irq.h>
#include <asm/delay.h>
#include <asm/clock.h>
-#include <asm/cpu/dac.h>
-#include <asm/cpu/timer.h>
+#include <cpu/dac.h>
#include <asm/machvec.h>
#include <mach/hp6xx.h>
#include <asm/hd64461.h>
#define MODNAME "sh_dac_audio"
-#define TMU_TOCR_INIT 0x00
-
-#define TMU1_TCR_INIT 0x0020 /* Clock/4, rising edge; interrupt on */
-#define TMU1_TSTR_INIT 0x02 /* Bit to turn on TMU1 */
-
#define BUFFER_SIZE 48000
static int rate;
static int empty;
static char *data_buffer, *buffer_begin, *buffer_end;
static int in_use, device_major;
+static struct hrtimer hrtimer;
+static ktime_t wakeups_per_second;
static void dac_audio_start_timer(void)
{
- u8 tstr;
-
- tstr = ctrl_inb(TMU_TSTR);
- tstr |= TMU1_TSTR_INIT;
- ctrl_outb(tstr, TMU_TSTR);
+ hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL);
}
static void dac_audio_stop_timer(void)
{
- u8 tstr;
-
- tstr = ctrl_inb(TMU_TSTR);
- tstr &= ~TMU1_TSTR_INIT;
- ctrl_outb(tstr, TMU_TSTR);
+ hrtimer_cancel(&hrtimer);
}
static void dac_audio_reset(void)
@@ -77,38 +66,30 @@ static void dac_audio_sync(void)
static void dac_audio_start(void)
{
if (mach_is_hp6xx()) {
- u16 v = inw(HD64461_GPADR);
+ u16 v = __raw_readw(HD64461_GPADR);
v &= ~HD64461_GPADR_SPEAKER;
- outw(v, HD64461_GPADR);
+ __raw_writew(v, HD64461_GPADR);
}
sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
- ctrl_outw(TMU1_TCR_INIT, TMU1_TCR);
}
static void dac_audio_stop(void)
{
dac_audio_stop_timer();
if (mach_is_hp6xx()) {
- u16 v = inw(HD64461_GPADR);
+ u16 v = __raw_readw(HD64461_GPADR);
v |= HD64461_GPADR_SPEAKER;
- outw(v, HD64461_GPADR);
+ __raw_writew(v, HD64461_GPADR);
}
- sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
+ sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
}
static void dac_audio_set_rate(void)
{
- unsigned long interval;
- struct clk *clk;
-
- clk = clk_get(NULL, "module_clk");
- interval = (clk_get_rate(clk) / 4) / rate;
- clk_put(clk);
- ctrl_outl(interval, TMU1_TCOR);
- ctrl_outl(interval, TMU1_TCNT);
+ wakeups_per_second = ktime_set(0, 1000000000 / rate);
}
static int dac_audio_ioctl(struct inode *inode, struct file *file,
@@ -265,32 +246,26 @@ const struct file_operations dac_audio_fops = {
.release = dac_audio_release,
};
-static irqreturn_t timer1_interrupt(int irq, void *dev)
+static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle)
{
- unsigned long timer_status;
-
- timer_status = ctrl_inw(TMU1_TCR);
- timer_status &= ~0x100;
- ctrl_outw(timer_status, TMU1_TCR);
-
if (!empty) {
sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL);
buffer_begin++;
if (buffer_begin == data_buffer + BUFFER_SIZE)
buffer_begin = data_buffer;
- if (buffer_begin == buffer_end) {
+ if (buffer_begin == buffer_end)
empty = 1;
- dac_audio_stop_timer();
- }
}
- return IRQ_HANDLED;
+
+ if (!empty)
+ hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL);
+
+ return HRTIMER_NORESTART;
}
static int __init dac_audio_init(void)
{
- int retval;
-
if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) {
printk(KERN_ERR "Cannot register dsp device");
return device_major;
@@ -306,21 +281,25 @@ static int __init dac_audio_init(void)
rate = 8000;
dac_audio_set_rate();
- retval =
- request_irq(TIMER1_IRQ, timer1_interrupt, IRQF_DISABLED, MODNAME, 0);
- if (retval < 0) {
- printk(KERN_ERR "sh_dac_audio: IRQ %d request failed\n",
- TIMER1_IRQ);
- return retval;
- }
+ /* Today: High Resolution Timer driven DAC playback.
+ * The timer callback gets called once per sample. Ouch.
+ *
+ * Future: A much better approach would be to use the
+ * SH7720 CMT+DMAC+DAC hardware combination like this:
+ * - Program sample rate using CMT0 or CMT1
+ * - Program DMAC to use CMT for timing and output to DAC
+ * - Play sound using DMAC, let CPU sleep.
+ * - While at it, rewrite this driver to use ALSA.
+ */
+
+ hrtimer_init(&hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ hrtimer.function = sh_dac_audio_timer;
return 0;
}
static void __exit dac_audio_exit(void)
{
- free_irq(TIMER1_IRQ, 0);
-
unregister_sound_dsp(device_major);
kfree((void *)data_buffer);
}
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 81bc93e5f1e..7337abdbe4e 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -958,10 +958,13 @@ static int patch_sigmatel_stac9708_3d(struct snd_ac97 * ac97)
}
static const struct snd_kcontrol_new snd_ac97_sigmatel_4speaker =
-AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch", AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+AC97_SINGLE("Sigmatel 4-Speaker Stereo Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 2, 1, 0);
+/* "Sigmatel " removed due to excessive name length: */
static const struct snd_kcontrol_new snd_ac97_sigmatel_phaseinvert =
-AC97_SINGLE("Sigmatel Surround Phase Inversion Playback Switch", AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
+AC97_SINGLE("Surround Phase Inversion Playback Switch",
+ AC97_SIGMATEL_DAC2INVERT, 3, 1, 0);
static const struct snd_kcontrol_new snd_ac97_sigmatel_controls[] = {
AC97_SINGLE("Sigmatel DAC 6dB Attenuate", AC97_SIGMATEL_ANALOG, 1, 1, 0),
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index ad2888705d2..c111efe61c3 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -800,7 +800,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
"Capture Volume",
"External Amplifier",
"Sigmatel 4-Speaker Stereo Playback Switch",
- "Sigmatel Surround Phase Inversion Playback ",
+ "Surround Phase Inversion Playback Switch",
NULL
};
static char *ca0106_rename_ctls[] = {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21e99cfa8c4..3128e1a6bc6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2141,6 +2141,7 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
/* forced codec slots */
+ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
{}
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 56ce19e68cb..4fcbe21829a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = {
static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
+ SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP),
SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
CXT5051_LAPTOP),
SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b8a0d3e7927..0fd258eba3a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -776,6 +776,12 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
if (pincap & AC_PINCAP_VREF_80)
val = PIN_VREF80;
+ else if (pincap & AC_PINCAP_VREF_50)
+ val = PIN_VREF50;
+ else if (pincap & AC_PINCAP_VREF_100)
+ val = PIN_VREF100;
+ else if (pincap & AC_PINCAP_VREF_GRD)
+ val = PIN_VREFGRD;
}
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
@@ -12058,6 +12064,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 03b3646018a..d2fd8ef6aef 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -150,6 +150,7 @@ enum {
STAC_D965_REF,
STAC_D965_3ST,
STAC_D965_5ST,
+ STAC_D965_5ST_NO_FP,
STAC_DELL_3ST,
STAC_DELL_BIOS,
STAC_927X_MODELS
@@ -2154,6 +2155,13 @@ static unsigned int d965_5st_pin_configs[14] = {
0x40000100, 0x40000100
};
+static unsigned int d965_5st_no_fp_pin_configs[14] = {
+ 0x40000100, 0x40000100, 0x0181304e, 0x01014010,
+ 0x01a19040, 0x01011012, 0x01016011, 0x40000100,
+ 0x40000100, 0x40000100, 0x40000100, 0x01442070,
+ 0x40000100, 0x40000100
+};
+
static unsigned int dell_3st_pin_configs[14] = {
0x02211230, 0x02a11220, 0x01a19040, 0x01114210,
0x01111212, 0x01116211, 0x01813050, 0x01112214,
@@ -2166,6 +2174,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
[STAC_D965_REF] = ref927x_pin_configs,
[STAC_D965_3ST] = d965_3st_pin_configs,
[STAC_D965_5ST] = d965_5st_pin_configs,
+ [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs,
[STAC_DELL_3ST] = dell_3st_pin_configs,
[STAC_DELL_BIOS] = NULL,
};
@@ -2176,6 +2185,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = {
[STAC_D965_REF] = "ref",
[STAC_D965_3ST] = "3stack",
[STAC_D965_5ST] = "5stack",
+ [STAC_D965_5ST_NO_FP] = "5stack-no-fp",
[STAC_DELL_3ST] = "dell-3stack",
[STAC_DELL_BIOS] = "dell-bios",
};
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 823296d7d57..a6b88482637 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -3347,7 +3347,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip,
[QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface,
[QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface,
[QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface,
- [QUIRK_MIDI_RAW] = snd_usb_create_midi_interface,
+ [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface,
[QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface,
[QUIRK_MIDI_CME] = snd_usb_create_midi_interface,
[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 36e4f7a29ad..8e7f78941ba 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -153,7 +153,7 @@ enum quirk_type {
QUIRK_MIDI_YAMAHA,
QUIRK_MIDI_MIDIMAN,
QUIRK_MIDI_NOVATION,
- QUIRK_MIDI_RAW,
+ QUIRK_MIDI_FASTLANE,
QUIRK_MIDI_EMAGIC,
QUIRK_MIDI_CME,
QUIRK_MIDI_US122L,
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 26bad373fe6..2fb35cc22a3 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1778,8 +1778,18 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip,
umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
- case QUIRK_MIDI_RAW:
+ case QUIRK_MIDI_FASTLANE:
umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
+ /*
+ * Interface 1 contains isochronous endpoints, but with the same
+ * numbers as in interface 0. Since it is interface 1 that the
+ * USB core has most recently seen, these descriptors are now
+ * associated with the endpoint numbers. This will foul up our
+ * attempts to submit bulk/interrupt URBs to the endpoints in
+ * interface 0, so we have to make sure that the USB core looks
+ * again at interface 0 by calling usb_set_interface() on it.
+ */
+ usb_set_interface(umidi->chip->dev, 0, 0);
err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
break;
case QUIRK_MIDI_EMAGIC:
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 647ef502965..5d955aaad85 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1868,7 +1868,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.data = & (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
- .type = QUIRK_MIDI_RAW
+ .type = QUIRK_MIDI_FASTLANE
},
{
.ifnum = 1,
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-02 09:33:51 +0000