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* [GIT PULL] ALSA fixes for 2.6.26-rc3
@ 2008-05-25 16:49 Takashi Iwai
  0 siblings, 0 replies; 2+ messages in thread
From: Takashi Iwai @ 2008-05-25 16:49 UTC (permalink / raw)
  To: Linus Torvalds; +Cc: linux-kernel

Hi Linus,

please pull another ALSA updates for 2.6.26-rc3 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

This includes a few trivial fixes for HD-audio and one fix for
snd-pcsp.


Thanks!

Takashi

===

Mauro Carvalho Chehab (1):
      [ALSA] hda - Fix capture mute Widget for stac9250/9251

Stas Sergeev (1):
      [ALSA] snd-pcsp - fix pcsp_treble_info() to honour an item number

Takashi Iwai (2):
      [ALSA] hda - Fix noise on VT1708 codec
      [ALSA] hda - Fix COEF and EAPD in ALC889 auto-configuration mode

Travis Place (2):
      [ALSA] hda - Add model for ASUS P5K-E/WIFI-AP
      [ALSA] hda - Added support for Foxconn P35AX-S mainboard


 sound/drivers/pcsp/pcsp.h       |    6 ++++--
 sound/drivers/pcsp/pcsp_mixer.c |    3 ++-
 sound/pci/hda/patch_analog.c    |    1 +
 sound/pci/hda/patch_realtek.c   |    3 +++
 sound/pci/hda/patch_sigmatel.c  |    2 +-
 sound/pci/hda/patch_via.c       |   20 ++++++++++++++++++++
 6 files changed, 31 insertions(+), 4 deletions(-)

---
diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h
index f07cc1e..1d661f7 100644
--- a/sound/drivers/pcsp/pcsp.h
+++ b/sound/drivers/pcsp/pcsp.h
@@ -24,7 +24,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 /* default timer freq for PC-Speaker: 18643 Hz */
 #define DIV_18KHZ 64
 #define MAX_DIV DIV_18KHZ
-#define CUR_DIV() (MAX_DIV >> chip->treble)
+#define CALC_DIV(d) (MAX_DIV >> (d))
+#define CUR_DIV() CALC_DIV(chip->treble)
 #define PCSP_MAX_TREBLE 1
 
 /* unfortunately, with hrtimers 37KHz does not work very well :( */
@@ -36,7 +37,8 @@ static DEFINE_SPINLOCK(i8253_lock);
 #define PCSP_DEFAULT_SDIV (DIV_18KHZ >> 1)
 #define PCSP_DEFAULT_SRATE (PIT_TICK_RATE / PCSP_DEFAULT_SDIV)
 #define PCSP_INDEX_INC() (1 << (PCSP_MAX_TREBLE - chip->treble))
-#define PCSP_RATE() (PIT_TICK_RATE / CUR_DIV())
+#define PCSP_CALC_RATE(i) (PIT_TICK_RATE / CALC_DIV(i))
+#define PCSP_RATE() PCSP_CALC_RATE(chip->treble)
 #define PCSP_MIN_RATE__1 MAX_DIV/PIT_TICK_RATE
 #define PCSP_MAX_RATE__1 MIN_DIV/PIT_TICK_RATE
 #define PCSP_MAX_PERIOD_NS (1000000000ULL * PCSP_MIN_RATE__1)
diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c
index 64a695f..caeb0f5 100644
--- a/sound/drivers/pcsp/pcsp_mixer.c
+++ b/sound/drivers/pcsp/pcsp_mixer.c
@@ -50,7 +50,8 @@ static int pcsp_treble_info(struct snd_kcontrol *kcontrol,
 	uinfo->value.enumerated.items = chip->max_treble + 1;
 	if (uinfo->value.enumerated.item > chip->max_treble)
 		uinfo->value.enumerated.item = chip->max_treble;
-	sprintf(uinfo->value.enumerated.name, "%d", PCSP_RATE());
+	sprintf(uinfo->value.enumerated.name, "%d",
+			PCSP_CALC_RATE(uinfo->value.enumerated.item));
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605a..ff1b922 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = {
 static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
 	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f5..8f31247 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -853,6 +853,7 @@ do_sku:
 		case 0x10ec0269:
 		case 0x10ec0862:
 		case 0x10ec0662:	
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x14, 0,
 					    AC_VERB_SET_EAPD_BTLENABLE, 2);
 			snd_hda_codec_write(codec, 0x15, 0,
@@ -877,6 +878,7 @@ do_sku:
 		case 0x10ec0883:
 		case 0x10ec0885:
 		case 0x10ec0888:
+		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_COEF_INDEX, 7);
 			tmp = snd_hda_codec_read(codec, 0x20, 0,
@@ -7743,6 +7745,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd..a4f44a0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
 	{ } /* end */
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81..e7e4352 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
 	},
 };
 
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x10, /* NID to query formats and rates */
+	/* We got noisy outputs on the right channel on VT1708 when
+	 * 24bit samples are used.  Until any workaround is found,
+	 * disable the 24bit format, so far.
+	 */
+	.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	.ops = {
+		.open = via_playback_pcm_open,
+		.prepare = via_playback_pcm_prepare,
+		.cleanup = via_playback_pcm_cleanup
+	},
+};
+
 static struct hda_pcm_stream vt1708_pcm_analog_capture = {
 	.substreams = 2,
 	.channels_min = 2,
@@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec)
 	
 	spec->stream_name_analog = "VT1708 Analog";
 	spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+	/* disable 32bit format on VT1708 */
+	if (codec->vendor_id == 0x11061708)
+		spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
 	spec->stream_analog_capture = &vt1708_pcm_analog_capture;
 
 	spec->stream_name_digital = "VT1708 Digital";

^ permalink raw reply	[flat|nested] 2+ messages in thread

* [GIT PULL] ALSA fixes for 2.6.26-rc3
@ 2008-05-19 16:04 Takashi Iwai
  0 siblings, 0 replies; 2+ messages in thread
From: Takashi Iwai @ 2008-05-19 16:04 UTC (permalink / raw)
  To: Linus Torvalds; +Cc: linux-kernel

Hi Linus,

please pull ALSA updates for 2.6.26-rc3 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

This contains a few trivial fixes for snd-pcsp driver.


Thanks!

Takashi


Stas Sergeev (4):
      snd-pcsp: adjust help texts to frighten users
      snd-pcsp: put back the compatibility code for the older alsa-libs
      snd-pcsp: depend on CONFIG_EXPERIMENTAL
      snd-pcsp: silent misleading warning


 sound/Kconfig                 |    5 -----
 sound/drivers/Kconfig         |   15 +++++++++++++--
 sound/drivers/pcsp/pcsp_lib.c |   23 +++++++++++++++++------
 3 files changed, 30 insertions(+), 13 deletions(-)

---
diff --git a/sound/Kconfig b/sound/Kconfig
index b2a2db4..4247406 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -28,11 +28,6 @@ config SOUND
 	  and read <file:Documentation/sound/oss/README.modules>; the module
 	  will be called soundcore.
 
-	  I'm told that even without a sound card, you can make your computer
-	  say more than an occasional beep, by programming the PC speaker.
-	  Kernel patches and supporting utilities to do that are in the pcsp
-	  package, available at <ftp://ftp.infradead.org/pub/pcsp/>.
-
 source "sound/oss/dmasound/Kconfig"
 
 if !M68K
diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig
index 379bcb0..602b58e 100644
--- a/sound/drivers/Kconfig
+++ b/sound/drivers/Kconfig
@@ -5,9 +5,10 @@ menu "Generic devices"
 
 
 config SND_PCSP
-	tristate "PC-Speaker support"
+	tristate "PC-Speaker support (READ HELP!)"
 	depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS
 	depends on INPUT
+	depends on EXPERIMENTAL
 	depends on SND
 	select SND_PCM
 	help
@@ -18,11 +19,21 @@ config SND_PCSP
 
 	  You can compile this as a module which will be called snd-pcsp.
 
+	  WARNING: if you already have a soundcard, enabling this
+	  driver may lead to a problem. Namely, it may get loaded
+	  before the other sound driver of yours, making the
+	  pc-speaker a default sound device. Which is likely not
+	  what you want. To make this driver play nicely with other
+	  sound driver, you can add this into your /etc/modprobe.conf:
+	  options snd-pcsp index=2
+
 	  You don't need this driver if you only want your pc-speaker to beep.
 	  You don't need this driver if you have a tablet piezo beeper
 	  in your PC instead of the real speaker.
 
-	  It should not hurt to say Y or M here in all other cases.
+	  Say N if you have a sound card.
+	  Say M if you don't.
+	  Say Y only if you really know what you do.
 
 config SND_MPU401_UART
         tristate
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index ac6238e..7ad4a15 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -18,6 +18,8 @@ module_param(nforce_wa, bool, 0444);
 MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
 		"(expect bad sound)");
 
+#define DMIX_WANTS_S16	1
+
 static void pcsp_start_timer(unsigned long dummy)
 {
 	hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL);
@@ -47,7 +49,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 {
 	unsigned long flags;
 	unsigned char timer_cnt, val;
-	int periods_elapsed;
+	int fmt_size, periods_elapsed;
 	u64 ns;
 	size_t period_bytes, buffer_bytes;
 	struct snd_pcm_substream *substream;
@@ -92,8 +94,11 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 		goto exit_nr_unlock2;
 
 	runtime = substream->runtime;
-	/* assume it is u8 mono */
-	val = runtime->dma_area[chip->playback_ptr];
+	fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3;
+	/* assume it is mono! */
+	val = runtime->dma_area[chip->playback_ptr + fmt_size - 1];
+	if (snd_pcm_format_signed(runtime->format))
+		val ^= 0x80;
 	timer_cnt = val * CUR_DIV() / 256;
 
 	if (timer_cnt && chip->enable) {
@@ -111,12 +116,14 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 
 	period_bytes = snd_pcm_lib_period_bytes(substream);
 	buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
-	chip->playback_ptr += PCSP_INDEX_INC();
+	chip->playback_ptr += PCSP_INDEX_INC() * fmt_size;
 	periods_elapsed = chip->playback_ptr - chip->period_ptr;
 	if (periods_elapsed < 0) {
-		printk(KERN_WARNING "PCSP: playback_ptr inconsistent "
+#if PCSP_DEBUG
+		printk(KERN_INFO "PCSP: buffer_bytes mod period_bytes != 0 ? "
 			"(%zi %zi %zi)\n",
 			chip->playback_ptr, period_bytes, buffer_bytes);
+#endif
 		periods_elapsed += buffer_bytes;
 	}
 	periods_elapsed /= period_bytes;
@@ -270,7 +277,11 @@ static struct snd_pcm_hardware snd_pcsp_playback = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED |
 		 SNDRV_PCM_INFO_HALF_DUPLEX |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
-	.formats = SNDRV_PCM_FMTBIT_U8,
+	.formats = (SNDRV_PCM_FMTBIT_U8
+#if DMIX_WANTS_S16
+		    | SNDRV_PCM_FMTBIT_S16_LE
+#endif
+	    ),
 	.rates = SNDRV_PCM_RATE_KNOT,
 	.rate_min = PCSP_DEFAULT_SRATE,
 	.rate_max = PCSP_DEFAULT_SRATE,

^ permalink raw reply	[flat|nested] 2+ messages in thread

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