November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 0/5] Support for OMAP4 Digital Microphone interface
by Peter Ujfalusi 28 Nov '11

28 Nov '11

Hello, The following series will add support for OMAP4 DMIC interface, and enable them on sdp4430/Blaze boards. Changes since v2: - Use module_platform_driver - convert to use devm_ - Add clk_id for the output clock - Extend the comment to explain why the driven can change the divider for 192KHz sampling rate. Changes since v1: - 192KHz rate support fixed (need to use different divider) - Hold reference for dmic_fclk over the lifetime of the driver - Removed the clkdiv callback, and replaced with frequency based configuration for the external components (dmic frequency) Regards, Peter --- Peter Ujfalusi (5): OMAP4: hwmod: Add names for DMIC memory address space ASoC: OMAP4: omap-dmic: Initial support for OMAP DMIC OMAP4: devices: Register OMAP4 DMIC platform device OMAP4: board-4430sdp: Register platform device for digimic codec ASoC: sdp4430: Add support for digital microphones arch/arm/mach-omap2/board-4430sdp.c | 6 + arch/arm/mach-omap2/devices.c | 22 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 + sound/soc/omap/Kconfig | 5 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-dmic.c | 511 ++++++++++++++++++++++++++++ sound/soc/omap/omap-dmic.h | 69 ++++ sound/soc/omap/sdp4430.c | 85 ++++- 8 files changed, 691 insertions(+), 11 deletions(-) create mode 100644 sound/soc/omap/omap-dmic.c create mode 100644 sound/soc/omap/omap-dmic.h -- 1.7.8.rc3

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Re: [alsa-devel] [PATCH v2] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 28 Nov '11

28 Nov '11

On Fri, Nov 25, 2011 at 09:39:03PM +0900, Tomoya MORINAGA wrote: This is better than the previous version but the major issue is that the driver isn't really fitting in with the subsystem - it's still relying on hard coded register write sequences for example. > ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high > resistance to voltage noise. On chip regulator realizes power supply rejection > ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/ You should fix the word wrapping in your commit message to be well within 80 columns. > +struct ml26124_priv { > + enum snd_soc_control_type control_type; Since the device only supports I2C you should just unconditionally use I2C. You should also use regmap for register access. > +static const struct snd_kcontrol_new ml26124_snd_controls[] = { > + SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0, 0xff, 1, rec_play_digi_vol), > + SOC_SINGLE("Digital Volume Switch", ML26124_FILTER_EN, 4, 1, 0), This needs either a Playback or Capture in there to match up with the above unless it genuinely does mute both which would be really weird. > + SND_SOC_DAPM_VMID("VMID"), Remove this unless it's actually used in your routing. > + SND_SOC_DAPM_MICBIAS("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0), This should be a supply widget, MICBIAS widgets are legacy. > + SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0x1f, 0, > + &ml26124_output_mixer_controls[0], > + ARRAY_SIZE(ml26124_output_mixer_controls)), Do you *really* have a shift of 0x1f for the enable bit? > +static int ml26124_snd_card_codec_set(int number, struct ml26124_priv *priv, > + struct snd_soc_codec *codec) > +{ > + unsigned char data; > + unsigned int rate = priv->rate; > + unsigned int channels = priv->ch; > + > + data = snd_soc_read(codec, ML26124_PW_MICBIAS_VOL); > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); /* soft reset assert */ > + snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); /* soft reset negate */ > + > + snd_soc_update_bits(codec, 0x0c, 0x1f, 0); /* Stop clock */ The same issue as with your previous submission applies to this function - you shouldn't have hard coded register write sequences in your code, the driver should use the facilities of the subsystem to allow the user to configure it at runtime. Given how big an issue this is I've stopped reading at this point.

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[alsa-devel] [PATCH] ASoC: Convert imx directory to module_platform_driver
by Axel Lin 28 Nov '11

28 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +----------- sound/soc/imx/imx-pcm-fiq.c | 12 +----------- sound/soc/imx/imx-ssi.c | 13 +------------ 3 files changed, 3 insertions(+), 34 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 7d28de9..aecdba9 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -326,16 +326,6 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 8df0fae2..d7ea0b3 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -331,14 +331,4 @@ static struct platform_driver imx_pcm_driver = { .remove = __devexit_p(imx_soc_platform_remove), }; -static int __init snd_imx_pcm_init(void) -{ - return platform_driver_register(&imx_pcm_driver); -} -module_init(snd_imx_pcm_init); - -static void __exit snd_imx_pcm_exit(void) -{ - platform_driver_unregister(&imx_pcm_driver); -} -module_exit(snd_imx_pcm_exit); +module_platform_driver(imx_pcm_driver); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 4c05e2b..d4b6367 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -757,18 +757,7 @@ static struct platform_driver imx_ssi_driver = { }, }; -static int __init imx_ssi_init(void) -{ - return platform_driver_register(&imx_ssi_driver); -} - -static void __exit imx_ssi_exit(void) -{ - platform_driver_unregister(&imx_ssi_driver); -} - -module_init(imx_ssi_init); -module_exit(imx_ssi_exit); +module_platform_driver(imx_ssi_driver); /* Module information */ MODULE_AUTHOR("Sascha Hauer, <s.hauer(a)pengutronix.de>"); -- 1.7.5.4

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[alsa-devel] snd_pcm_drain and partial periods
by Trent Piepho 28 Nov '11

28 Nov '11

Say an application does not write a full multiple of the period size with snd_pcm_writei(). It's playing a clip and the clip ends and it doesn't happen to end on a multiple of 6000 frames or whatever the period size is. Now the application calls snd_pcm_drain() to wait for the clip to finish. How does the driver know to stop when it gets to the end of the supplied data? I don't see anywhere where this will get taken care of. It seems like the driver will keep playing until it calls snd_pcm_period_elapsed() at the end of the final period. Then the alsa pcm layer will call the ->trigger() method and stop the stream. Hopefully before the driver has started the next DMA for the next period! The application only wrote data for part of the final period, so the driver will have played past the end of the supplied data. I don't see any documentation that says you must always supply a multiple of the period size with snd_pcm_write[in](). Nor about how to deal with non-blocking mode and only part of the data getting written. I don't see any documentation for snd_pcm_drain() that says it will keep playing until it finishes a period, even if the data runs out before hand. "For playback wait for all pending frames to be played and then stop the PCM." Seems pretty clear that only the pending frames are played and not the rest of a period. Yet the driver I'm working on definitely keeps playing to the end of the period on drain. Could be a driver problem, but I don't see how any other drivers do anything differently here.

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[alsa-devel] [PATCH] ASoC: Convert WM8962 to devm_kzalloc()
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e01232f..1ec7271 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4182,7 +4182,8 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, unsigned int reg; int ret, i; - wm8962 = kzalloc(sizeof(struct wm8962_priv), GFP_KERNEL); + wm8962 = devm_kzalloc(&i2c->dev, sizeof(struct wm8962_priv), + GFP_KERNEL); if (wm8962 == NULL) return -ENOMEM; @@ -4199,7 +4200,7 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, wm8962->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); - goto err_alloc; + goto err; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), @@ -4273,8 +4274,7 @@ err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); err_get: regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); -err_alloc: - kfree(wm8962); +err: return ret; } @@ -4285,7 +4285,6 @@ static __devexit int wm8962_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8962->regmap); regulator_bulk_free(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Log automatic pin disconnection per CODEC rather than per card
by Mark Brown 27 Nov '11

27 Nov '11

This makes the output a bit less confusing on multi-CODEC systems as the same pin may appear in multiple CODECs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 81f91f1..c3cde81 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3002,7 +3002,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; - dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n", + dev_dbg(codec->dev, "Auto NC: DAPMs: card:%p codec:%p\n", &card->dapm, &codec->dapm); list_for_each_entry(w, &card->widgets, list) { @@ -3012,10 +3012,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec) case snd_soc_dapm_input: case snd_soc_dapm_output: case snd_soc_dapm_micbias: - dev_dbg(card->dev, "Auto NC: Checking widget %s\n", + dev_dbg(codec->dev, "Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { - dev_dbg(card->dev, + dev_dbg(codec->dev, "... Not in map; disabling\n"); snd_soc_dapm_nc_pin(dapm, w->name); } -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Fix CODEC enumeration for auto_nc_codec_pins
by Mark Brown 27 Nov '11

27 Nov '11

We need to enumerate all the CODECs that are part of the card we're instantiating, not all the CODECs that are in the system as the system may have multiple cards. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 169ab29..e0f4515 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1489,7 +1489,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_widgets(&card->dapm); if (card->fully_routed) - list_for_each_entry(codec, &codec_list, list) + list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); ret = snd_card_register(card->snd_card); -- 1.7.7.3

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[alsa-devel] [PATCH] ALSA: convert sound/* to use module_platform_driver()
by Axel Lin 27 Nov '11

27 Nov '11

This patch converts the drivers in sound/* to use the module_platform_driver() macro which makes the code smaller and a bit simpler. Cc: Joachim Foerster <JOFT(a)gmx.de> Cc: Rafael Ignacio Zurita <rizurita(a)yahoo.com> Cc: David S. Miller <davem(a)davemloft.net> Cc: Martin Habets <mhabets(a)users.sourceforge.net> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/arm/pxa2xx-ac97.c | 13 +------------ sound/drivers/ml403-ac97cr.c | 13 +------------ sound/sh/sh_dac_audio.c | 13 +------------ sound/sparc/cs4231.c | 13 +------------ sound/sparc/dbri.c | 14 +------------- 5 files changed, 5 insertions(+), 61 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d94118..3a39626 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -251,18 +251,7 @@ static struct platform_driver pxa2xx_ac97_driver = { }, }; -static int __init pxa2xx_ac97_init(void) -{ - return platform_driver_register(&pxa2xx_ac97_driver); -} - -static void __exit pxa2xx_ac97_exit(void) -{ - platform_driver_unregister(&pxa2xx_ac97_driver); -} - -module_init(pxa2xx_ac97_init); -module_exit(pxa2xx_ac97_exit); +module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 2ee82c5..07ede97 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1341,15 +1341,4 @@ static struct platform_driver snd_ml403_ac97cr_driver = { }, }; -static int __init alsa_card_ml403_ac97cr_init(void) -{ - return platform_driver_register(&snd_ml403_ac97cr_driver); -} - -static void __exit alsa_card_ml403_ac97cr_exit(void) -{ - platform_driver_unregister(&snd_ml403_ac97cr_driver); -} - -module_init(alsa_card_ml403_ac97cr_init) -module_exit(alsa_card_ml403_ac97cr_exit) +module_platform_driver(snd_ml403_ac97cr_driver); diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 56bcb46..b11f82b 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -441,15 +441,4 @@ static struct platform_driver driver = { }, }; -static int __init sh_dac_init(void) -{ - return platform_driver_register(&driver); -} - -static void __exit sh_dac_exit(void) -{ - platform_driver_unregister(&driver); -} - -module_init(sh_dac_init); -module_exit(sh_dac_exit); +module_platform_driver(driver); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 0e618f8..9aa90e0 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2118,15 +2118,4 @@ static struct platform_driver cs4231_driver = { .remove = __devexit_p(cs4231_remove), }; -static int __init cs4231_init(void) -{ - return platform_driver_register(&cs4231_driver); -} - -static void __exit cs4231_exit(void) -{ - platform_driver_unregister(&cs4231_driver); -} - -module_init(cs4231_init); -module_exit(cs4231_exit); +module_platform_driver(cs4231_driver); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 4a4f1d7..6afe087 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2697,16 +2697,4 @@ static struct platform_driver dbri_sbus_driver = { .remove = __devexit_p(dbri_remove), }; -/* Probe for the dbri chip and then attach the driver. */ -static int __init dbri_init(void) -{ - return platform_driver_register(&dbri_sbus_driver); -} - -static void __exit dbri_exit(void) -{ - platform_driver_unregister(&dbri_sbus_driver); -} - -module_init(dbri_init); -module_exit(dbri_exit); +module_platform_driver(dbri_sbus_driver); -- 1.7.5.4

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[alsa-devel] [PATCH 1/3] ASoC: Convert wm1250-ev1 driver to use devm_kzalloc()
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index cd0ec0f..aefb4f8 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -116,7 +116,7 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) if (!pdata) return 0; - wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); if (!wm1250) { dev_err(&i2c->dev, "Unable to allocate private data\n"); ret = -ENOMEM; @@ -134,15 +134,13 @@ static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); if (ret != 0) { dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); - goto err_alloc; + goto err; } dev_set_drvdata(&i2c->dev, wm1250); return ret; -err_alloc: - kfree(wm1250); err: return ret; } @@ -151,10 +149,8 @@ static void wm1250_ev1_free(struct i2c_client *i2c) { struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); - if (wm1250) { + if (wm1250) gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); - kfree(wm1250); - } } static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Correct name of Speyside Main Speaker widget
by Mark Brown 27 Nov '11

27 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 1925316..e0f55cb5 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -164,7 +164,7 @@ static int speyside_late_probe(struct snd_soc_card *card) snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); -- 1.7.7.3

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