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[alsa-devel] Question about snd_pcm_update_hw_ptr_post()
by Timur Tabi 04 Mar '08

04 Mar '08

Can someone explain to me this snippet from snd_pcm_update_hw_ptr_post(): if (avail >= runtime->stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); return -EPIPE; } During playback, avail = snd_pcm_playback_avail(). Does this mean that avail is number of frames available in the playback DMA buffer for new data? If so, why is avail >= runtime->stop_threshold considered an underrun condition? Does ALSA try to keep the amount of available data to a minimum? If so, how can I adjust runtime->stop_threshold? I'd like to see what happens if I make stop_threshold equal to the size of my DMA buffer. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 04 Mar '08

04 Mar '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

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[alsa-devel] [PATCH] Add Invert Switch for ROUT2
by Mark Brown 03 Mar '08

03 Mar '08

From: Graeme Gregory <graeme(a)openmoko.org> GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory <graeme(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0288275..76a5c7b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ -- 1.5.4.3

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[alsa-devel] ALSA C++ API updated for 1.0.16
by Lasse Kärkkäinen 01 Mar '08

01 Mar '08

Here's an updated version of the C++ wrapper that I posted earlier in September 2007. I would like this to be included in the ALSA distribution after some peer review. This wrapper has been used in UltraStar NG (a game shipped by Gentoo and others) since September, with no known issues other than that it is incomplete (very PCM-centric). As a reminder, the API closely follows the C API, and also allows mixing direct C API calls with itself transparently.

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[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 29 Feb '08

29 Feb '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

1 0

[alsa-devel] [PATCH] hda: incorrect DAC mixer on 92hd71bxxx
by Matthew Ranostay 29 Feb '08

29 Feb '08

If there was two line_outs[] it would overrun into the dmic nid struct below and create a phantom "Surround" mixer. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 16:52:43 2008 -0500 @@ -246,8 +246,8 @@ static hda_nid_t stac92hd71bxx_dmux_nids 0x1c, }; -static hda_nid_t stac92hd71bxx_dac_nids[1] = { - 0x10, /*0x11, */ +static hda_nid_t stac92hd71bxx_dac_nids[2] = { + 0x10, 0x11 }; #define STAC92HD71BXX_NUM_DMICS 2 @@ -2345,10 +2345,10 @@ static int stac92xx_auto_create_multi_ou for (i = 0; i < cfg->line_outs; i++) { - if (!spec->multiout.dac_nids[i]) + + nid = spec->multiout.dac_nids[i]; + if (!nid || nid == spec->multiout.hp_nid) continue; - - nid = spec->multiout.dac_nids[i]; if (i == 2) { /* Center/LFE */ @@ -3550,7 +3550,7 @@ again: spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->multiout.num_dacs = 1; + spec->multiout.num_dacs = ARRAY_SIZE(stac92hd71bxx_dac_nids); spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids;

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[alsa-devel] [PATCH] hda: 92HD
by Matthew Ranostay 29 Feb '08

29 Feb '08

Added core_init[] for several 92hd73xxx laptops. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 21:14:39 2008 -0500 @@ -525,6 +525,24 @@ static struct hda_verb stac92hd73xx_6ch_ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb dell_m6_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, /* setup import muxs */ { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -3437,6 +3455,7 @@ again: switch (spec->board_config) { case STAC_DELL_M6: + spec->init = dell_m6_core_init; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f:

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[alsa-devel] [PATCH] IEC958 definitions for consumer status channel, byte 4
by Pawel MOLL 29 Feb '08

29 Feb '08

Hi, Attached are patches for alsa-kernel and alsa-lib, adding mentioned definitions. Regards Paweł

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[alsa-devel] [PATCH] hda: disable power management on fixed ports.
by Matthew Ranostay 29 Feb '08

29 Feb '08

Power management can't be enabled on fixed ports, since the presence will always return false and prevent output. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 54a51b975975 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Feb 28 12:46:32 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Feb 28 20:39:28 2008 -0500 @@ -3015,11 +3015,15 @@ static int stac92xx_init(struct hda_code ? STAC_HP_EVENT : STAC_PWR_EVENT; int pinctl = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) + continue; + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26);

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[alsa-devel] [PATCH] hda/patch_sigmatel.c Panasonic CF-74 speaker output
by Kyler Laird 29 Feb '08

29 Feb '08

This patch causes ALSA to send sound to the speakers of a Panasonic CF-74 (Toughbook) instead of just its headphone jack. --kyler diff -Naur linux-2.6.24.1/sound/pci/hda/patch_sigmatel.c linux-2.6.24.1-ALSA_CF74/sound/pci/hda/patch_sigmatel.c --- linux-2.6.24.1/sound/pci/hda/patch_sigmatel.c 2008-02-08 14:55:30.000000000 -0500 +++ linux-2.6.24.1-ALSA_CF74/sound/pci/hda/patch_sigmatel.c 2008-02-11 12:05:11.824064617 -0500 @@ -2467,6 +2467,11 @@ spec->init = stac9200_core_init; spec->mixer = stac9200_mixer; + /* Kyler - fix for CF-74 speaker */ + spec->gpio_mask = 0x00000009; + spec->gpio_data = 0x00000000; + stac92xx_enable_gpio_mask(codec); + err = stac9200_parse_auto_config(codec); if (err < 0) { stac92xx_free(codec);

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