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

[alsa-devel] [PATCH] ASoC: convert sound/soc/codecs/* to use module_platform_driver()
by Axel Lin 29 Nov '11

29 Nov '11

This patch converts the drivers in sound/soc/codecs/* to use the module_platform_driver() macro which makes the code smaller and a bit simpler. Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Bas Vermeulen <bas.vermeulen(a)novero.com> Cc: "Manjunathappa, Prakash" <prakash.pm(a)ti.com> Cc: Misael Lopez Cruz <misael.lopez(a)ti.com> Cc: Vinod Koul <vinod.koul(a)linux.intel.com> Cc: Stephen Warren <swarren(a)nvidia.com> Cc: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Cc: Marek Belisko <marek.belisko(a)open-nandra.com> Cc: "Matti J. Aaltonen" <matti.j.aaltonen(a)nokia.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 12 +----------- sound/soc/codecs/ac97.c | 12 +----------- sound/soc/codecs/ad1980.c | 12 +----------- sound/soc/codecs/ad73311.c | 12 +----------- sound/soc/codecs/ads117x.c | 12 +----------- sound/soc/codecs/cq93vc.c | 12 +----------- sound/soc/codecs/cx20442.c | 12 +----------- sound/soc/codecs/dfbmcs320.c | 12 +----------- sound/soc/codecs/dmic.c | 15 ++------------- sound/soc/codecs/jz4740.c | 12 +----------- sound/soc/codecs/pcm3008.c | 15 ++------------- sound/soc/codecs/sn95031.c | 14 +------------- sound/soc/codecs/spdif_transciever.c | 13 +------------ sound/soc/codecs/stac9766.c | 12 +----------- sound/soc/codecs/twl4030.c | 15 ++------------- sound/soc/codecs/twl6040.c | 12 +----------- sound/soc/codecs/uda134x.c | 12 +----------- sound/soc/codecs/wl1273.c | 15 ++------------- sound/soc/codecs/wm8350.c | 12 +----------- sound/soc/codecs/wm8400.c | 12 +----------- sound/soc/codecs/wm8727.c | 12 +----------- sound/soc/codecs/wm8782.c | 12 +----------- sound/soc/codecs/wm8994.c | 13 +------------ sound/soc/codecs/wm9705.c | 12 +----------- sound/soc/codecs/wm9712.c | 12 +----------- sound/soc/codecs/wm9713.c | 12 +----------- 26 files changed, 30 insertions(+), 298 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index ea305b8..2d39123 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1481,17 +1481,7 @@ static struct platform_driver pm860x_codec_driver = { .remove = __devexit_p(pm860x_codec_remove), }; -static __init int pm860x_init(void) -{ - return platform_driver_register(&pm860x_codec_driver); -} -module_init(pm860x_init); - -static __exit void pm860x_exit(void) -{ - platform_driver_unregister(&pm860x_codec_driver); -} -module_exit(pm860x_exit); +module_platform_driver(pm860x_codec_driver); MODULE_DESCRIPTION("ASoC 88PM860x driver"); MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8f32167..221ec29 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -148,17 +148,7 @@ static struct platform_driver ac97_codec_driver = { .remove = __devexit_p(ac97_remove), }; -static int __init ac97_init(void) -{ - return platform_driver_register(&ac97_codec_driver); -} -module_init(ac97_init); - -static void __exit ac97_exit(void) -{ - platform_driver_unregister(&ac97_codec_driver); -} -module_exit(ac97_exit); +module_platform_driver(ac97_codec_driver); MODULE_DESCRIPTION("Soc Generic AC97 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index e3931cc..9bba7f8 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -277,17 +277,7 @@ static struct platform_driver ad1980_codec_driver = { .remove = __devexit_p(ad1980_remove), }; -static int __init ad1980_init(void) -{ - return platform_driver_register(&ad1980_codec_driver); -} -module_init(ad1980_init); - -static void __exit ad1980_exit(void) -{ - platform_driver_unregister(&ad1980_codec_driver); -} -module_exit(ad1980_exit); +module_platform_driver(ad1980_codec_driver); MODULE_DESCRIPTION("ASoC ad1980 driver (Obsolete)"); MODULE_AUTHOR("Roy Huang, Cliff Cai"); diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 8d793e9..ee7a68d 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -63,17 +63,7 @@ static struct platform_driver ad73311_codec_driver = { .remove = __devexit_p(ad73311_remove), }; -static int __init ad73311_init(void) -{ - return platform_driver_register(&ad73311_codec_driver); -} -module_init(ad73311_init); - -static void __exit ad73311_exit(void) -{ - platform_driver_unregister(&ad73311_codec_driver); -} -module_exit(ad73311_exit); +module_platform_driver(ad73311_codec_driver); MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 9082e0f..8103b93 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -58,17 +58,7 @@ static struct platform_driver ads117x_codec_driver = { .remove = __devexit_p(ads117x_remove), }; -static int __init ads117x_init(void) -{ - return platform_driver_register(&ads117x_codec_driver); -} -module_init(ads117x_init); - -static void __exit ads117x_exit(void) -{ - platform_driver_unregister(&ads117x_codec_driver); -} -module_exit(ads117x_exit); +module_platform_driver(ads117x_codec_driver); MODULE_DESCRIPTION("ASoC ads117x driver"); MODULE_AUTHOR("Graeme Gregory"); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index cbb3028..4854b47 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -206,17 +206,7 @@ static struct platform_driver cq93vc_codec_driver = { .remove = __devexit_p(cq93vc_platform_remove), }; -static int __init cq93vc_init(void) -{ - return platform_driver_register(&cq93vc_codec_driver); -} -module_init(cq93vc_init); - -static void __exit cq93vc_exit(void) -{ - platform_driver_unregister(&cq93vc_codec_driver); -} -module_exit(cq93vc_exit); +module_platform_driver(cq93vc_codec_driver); MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); MODULE_AUTHOR("Miguel Aguilar"); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..ae55e31 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -391,17 +391,7 @@ static struct platform_driver cx20442_platform_driver = { .remove = __exit_p(cx20442_platform_remove), }; -static int __init cx20442_init(void) -{ - return platform_driver_register(&cx20442_platform_driver); -} -module_init(cx20442_init); - -static void __exit cx20442_exit(void) -{ - platform_driver_unregister(&cx20442_platform_driver); -} -module_exit(cx20442_exit); +module_platform_driver(cx20442_platform_driver); MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); MODULE_AUTHOR("Janusz Krzysztofik"); diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c index 704bbde..bfe46aa 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/dfbmcs320.c @@ -55,17 +55,7 @@ static struct platform_driver dfmcs320_driver = { .remove = __devexit_p(dfbmcs320_remove), }; -static int __init dfbmcs320_init(void) -{ - return platform_driver_register(&dfmcs320_driver); -} -module_init(dfbmcs320_init); - -static void __exit dfbmcs320_exit(void) -{ - platform_driver_unregister(&dfmcs320_driver); -} -module_exit(dfbmcs320_exit); +module_platform_driver(dfmcs320_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 6fae765..9ec214e 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -78,8 +78,6 @@ static int __devexit dmic_dev_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:dmic-codec"); - static struct platform_driver dmic_driver = { .driver = { .name = "dmic-codec", @@ -89,18 +87,9 @@ static struct platform_driver dmic_driver = { .remove = __devexit_p(dmic_dev_remove), }; -static int __init dmic_init(void) -{ - return platform_driver_register(&dmic_driver); -} -module_init(dmic_init); - -static void __exit dmic_exit(void) -{ - platform_driver_unregister(&dmic_driver); -} -module_exit(dmic_exit); +module_platform_driver(dmic_driver); MODULE_DESCRIPTION("Generic DMIC driver"); MODULE_AUTHOR("Liam Girdwood <lrg(a)slimlogic.co.uk>"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:dmic-codec"); diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 64a479c..4fca8bc 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -424,17 +424,7 @@ static struct platform_driver jz4740_codec_driver = { }, }; -static int __init jz4740_codec_init(void) -{ - return platform_driver_register(&jz4740_codec_driver); -} -module_init(jz4740_codec_init); - -static void __exit jz4740_codec_exit(void) -{ - platform_driver_unregister(&jz4740_codec_driver); -} -module_exit(jz4740_codec_exit); +module_platform_driver(jz4740_codec_driver); MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f731651..dffab12 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -161,8 +161,6 @@ static int __devexit pcm3008_codec_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:pcm3008-codec"); - static struct platform_driver pcm3008_codec_driver = { .probe = pcm3008_codec_probe, .remove = __devexit_p(pcm3008_codec_remove), @@ -172,18 +170,9 @@ static struct platform_driver pcm3008_codec_driver = { }, }; -static int __init pcm3008_modinit(void) -{ - return platform_driver_register(&pcm3008_codec_driver); -} -module_init(pcm3008_modinit); - -static void __exit pcm3008_exit(void) -{ - platform_driver_unregister(&pcm3008_codec_driver); -} -module_exit(pcm3008_exit); +module_platform_driver(pcm3008_codec_driver); MODULE_DESCRIPTION("Soc PCM3008 driver"); MODULE_AUTHOR("Hugo Villeneuve"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pcm3008-codec"); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 65f2ef9..f99baa0 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -920,19 +920,7 @@ static struct platform_driver sn95031_codec_driver = { .remove = __devexit_p(sn95031_device_remove), }; -static int __init sn95031_init(void) -{ - pr_debug("driver init called\n"); - return platform_driver_register(&sn95031_codec_driver); -} -module_init(sn95031_init); - -static void __exit sn95031_exit(void) -{ - pr_debug("driver exit called\n"); - platform_driver_unregister(&sn95031_codec_driver); -} -module_exit(sn95031_exit); +module_platform_driver(sn95031_codec_driver); MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 6a1a7e7..112a49d 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -61,18 +61,7 @@ static struct platform_driver spdif_dit_driver = { }, }; -static int __init dit_modinit(void) -{ - return platform_driver_register(&spdif_dit_driver); -} - -static void __exit dit_exit(void) -{ - platform_driver_unregister(&spdif_dit_driver); -} - -module_init(dit_modinit); -module_exit(dit_exit); +module_platform_driver(spdif_dit_driver); MODULE_AUTHOR("Steve Chen <schen(a)mvista.com>"); MODULE_DESCRIPTION("SPDIF dummy codec driver"); diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index e4783a4..5581953 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -408,17 +408,7 @@ static struct platform_driver stac9766_codec_driver = { .remove = __devexit_p(stac9766_remove), }; -static int __init stac9766_init(void) -{ - return platform_driver_register(&stac9766_codec_driver); -} -module_init(stac9766_init); - -static void __exit stac9766_exit(void) -{ - platform_driver_unregister(&stac9766_codec_driver); -} -module_exit(stac9766_exit); +module_platform_driver(stac9766_codec_driver); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a3a528..108e596 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2283,8 +2283,6 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:twl4030-codec"); - static struct platform_driver twl4030_codec_driver = { .probe = twl4030_codec_probe, .remove = __devexit_p(twl4030_codec_remove), @@ -2294,18 +2292,9 @@ static struct platform_driver twl4030_codec_driver = { }, }; -static int __init twl4030_modinit(void) -{ - return platform_driver_register(&twl4030_codec_driver); -} -module_init(twl4030_modinit); - -static void __exit twl4030_exit(void) -{ - platform_driver_unregister(&twl4030_codec_driver); -} -module_exit(twl4030_exit); +module_platform_driver(twl4030_codec_driver); MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:twl4030-codec"); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 17930ed..a4a65dc 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1620,17 +1620,7 @@ static struct platform_driver twl6040_codec_driver = { .remove = __devexit_p(twl6040_codec_remove), }; -static int __init twl6040_codec_init(void) -{ - return platform_driver_register(&twl6040_codec_driver); -} -module_init(twl6040_codec_init); - -static void __exit twl6040_codec_exit(void) -{ - platform_driver_unregister(&twl6040_codec_driver); -} -module_exit(twl6040_codec_exit); +module_platform_driver(twl6040_codec_driver); MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); MODULE_AUTHOR("Misael Lopez Cruz"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 486aef6..d0f9d90 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -625,17 +625,7 @@ static struct platform_driver uda134x_codec_driver = { .remove = __devexit_p(uda134x_codec_remove), }; -static int __init uda134x_codec_init(void) -{ - return platform_driver_register(&uda134x_codec_driver); -} -module_init(uda134x_codec_init); - -static void __exit uda134x_codec_exit(void) -{ - platform_driver_unregister(&uda134x_codec_driver); -} -module_exit(uda134x_codec_exit); +module_platform_driver(uda134x_codec_driver); MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 9531c35..49fd302 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -499,8 +499,6 @@ static int __devexit wl1273_platform_remove(struct platform_device *pdev) return 0; } -MODULE_ALIAS("platform:wl1273-codec"); - static struct platform_driver wl1273_platform_driver = { .driver = { .name = "wl1273-codec", @@ -510,18 +508,9 @@ static struct platform_driver wl1273_platform_driver = { .remove = __devexit_p(wl1273_platform_remove), }; -static int __init wl1273_init(void) -{ - return platform_driver_register(&wl1273_platform_driver); -} -module_init(wl1273_init); - -static void __exit wl1273_exit(void) -{ - platform_driver_unregister(&wl1273_platform_driver); -} -module_exit(wl1273_exit); +module_platform_driver(wl1273_platform_driver); MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen(a)nokia.com>"); MODULE_DESCRIPTION("ASoC WL1273 codec driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wl1273-codec"); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3b846c9..3f1ed5f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1711,17 +1711,7 @@ static struct platform_driver wm8350_codec_driver = { .remove = __devexit_p(wm8350_remove), }; -static __init int wm8350_init(void) -{ - return platform_driver_register(&wm8350_codec_driver); -} -module_init(wm8350_init); - -static __exit void wm8350_exit(void) -{ - platform_driver_unregister(&wm8350_codec_driver); -} -module_exit(wm8350_exit); +module_platform_driver(wm8350_codec_driver); MODULE_DESCRIPTION("ASoC WM8350 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 07d84a8..a1173eb 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1477,17 +1477,7 @@ static struct platform_driver wm8400_codec_driver = { .remove = __devexit_p(wm8400_remove), }; -static __init int wm8400_init(void) -{ - return platform_driver_register(&wm8400_codec_driver); -} -module_init(wm8400_init); - -static __exit void wm8400_exit(void) -{ - platform_driver_unregister(&wm8400_codec_driver); -} -module_exit(wm8400_exit); +module_platform_driver(wm8400_codec_driver); MODULE_DESCRIPTION("ASoC WM8400 driver"); MODULE_AUTHOR("Mark Brown"); diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 7488082..fad90a3 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -67,17 +67,7 @@ static struct platform_driver wm8727_codec_driver = { .remove = __devexit_p(wm8727_remove), }; -static int __init wm8727_init(void) -{ - return platform_driver_register(&wm8727_codec_driver); -} -module_init(wm8727_init); - -static void __exit wm8727_exit(void) -{ - platform_driver_unregister(&wm8727_codec_driver); -} -module_exit(wm8727_exit); +module_platform_driver(wm8727_codec_driver); MODULE_DESCRIPTION("ASoC wm8727 driver"); MODULE_AUTHOR("Neil Jones"); diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f2ced71..3fdea98 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -63,17 +63,7 @@ static struct platform_driver wm8782_codec_driver = { .remove = __devexit_p(wm8782_remove), }; -static int __init wm8782_init(void) -{ - return platform_driver_register(&wm8782_codec_driver); -} -module_init(wm8782_init); - -static void __exit wm8782_exit(void) -{ - platform_driver_unregister(&wm8782_codec_driver); -} -module_exit(wm8782_exit); +module_platform_driver(wm8782_codec_driver); MODULE_DESCRIPTION("ASoC WM8782 driver"); MODULE_AUTHOR("Johannes Stezenbach <js(a)sig21.net>"); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 73db980..380e3f2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3579,18 +3579,7 @@ static struct platform_driver wm8994_codec_driver = { .remove = __devexit_p(wm8994_remove), }; -static __init int wm8994_init(void) -{ - return platform_driver_register(&wm8994_codec_driver); -} -module_init(wm8994_init); - -static __exit void wm8994_exit(void) -{ - platform_driver_unregister(&wm8994_codec_driver); -} -module_exit(wm8994_exit); - +module_platform_driver(wm8994_codec_driver); MODULE_DESCRIPTION("ASoC WM8994 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index edf6032..b720a43 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,17 +406,7 @@ static struct platform_driver wm9705_codec_driver = { .remove = __devexit_p(wm9705_remove), }; -static int __init wm9705_init(void) -{ - return platform_driver_register(&wm9705_codec_driver); -} -module_init(wm9705_init); - -static void __exit wm9705_exit(void) -{ - platform_driver_unregister(&wm9705_codec_driver); -} -module_exit(wm9705_exit); +module_platform_driver(wm9705_codec_driver); MODULE_DESCRIPTION("ASoC WM9705 driver"); MODULE_AUTHOR("Ian Molton"); diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index fd18127..4ce73f5 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -694,17 +694,7 @@ static struct platform_driver wm9712_codec_driver = { .remove = __devexit_p(wm9712_remove), }; -static int __init wm9712_init(void) -{ - return platform_driver_register(&wm9712_codec_driver); -} -module_init(wm9712_init); - -static void __exit wm9712_exit(void) -{ - platform_driver_unregister(&wm9712_codec_driver); -} -module_exit(wm9712_exit); +module_platform_driver(wm9712_codec_driver); MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); MODULE_AUTHOR("Liam Girdwood"); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 09360b6..edb5981 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1277,17 +1277,7 @@ static struct platform_driver wm9713_codec_driver = { .remove = __devexit_p(wm9713_remove), }; -static int __init wm9713_init(void) -{ - return platform_driver_register(&wm9713_codec_driver); -} -module_init(wm9713_init); - -static void __exit wm9713_exit(void) -{ - platform_driver_unregister(&wm9713_codec_driver); -} -module_exit(wm9713_exit); +module_platform_driver(wm9713_codec_driver); MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); MODULE_AUTHOR("Liam Girdwood"); -- 1.7.5.4

4 3

[alsa-devel] how to handle buffer underrun in a softsynth
by Henning Thielemann 29 Nov '11

29 Nov '11

I have written a software synthesizer in Haskell that fetches MIDI events from ALSA sequencer and plays the rendered signal via ALSA pcm. However it cannot cope correctly with buffer underruns. I tried to learn the correct way from the miniFMsynth example of http://alsamodular.sourceforge.net/alsa_programming_howto.html however this program floods my terminal with buffer underrun messages without even being connected to any MIDI event source. This is independent from how large I choose the BUFSIZE constant. I use: $ pkg-config --modversion alsa 1.0.14 Also, as far as I understand, that program allows to start notes only at block boundaries. This way it can ignore the timestamps of the incoming events. But this is crucial for me: I want to allow large blocksizes and thus I must allow notes to start anywhere inside a block. Now when I detect a buffer underrun on writing a block to an ALSA pcm sink, I like to know the time gap in order to adjust the timestamps of the incoming MIDI events. My question in short is: Is there an example software synthesizer as simple as miniFMsynth, that works (on my machine), supports note start within a block and copes with buffer underruns in a reasonable way? If no working example, is there some documentation on how this is intended to work? Thank you for your help Henning

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[alsa-devel] [PATCH] ALSA: hda_intel - revert a quirk that breaks all VIA chipsets
by Marc Vertes 29 Nov '11

29 Nov '11

This quirk sould be reverted. It has the following probems: 1) It breaks the default setup of all VIA chipset models (vendor 0x1106, product 0x3288), at least a dozen models are concerned, from VIA-VT82xx to VIA-VT17xx and VT20xx. All those chips rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 2) The string "ASUS MV2-MX SE" refers to a motherboard, not a sound chipset family, which is misleading on the impact of the quirk. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.h… Thanks, Marc -- Signed-of-by: Marc Vertes <mvertes(a)sigfox.com> --- linux-3.2-rc3/sound/pci/hda/hda_intel.c.orig 2011-11-24 05:20:28.000000000 +0100 +++ linux-3.2-rc3/sound/pci/hda/hda_intel.c 2011-11-28 17:29:01.044095455 +0100 @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),

2 1

[PATCH] ALSA: HDA: Fix Realtek automute Line Out -> Internal Speaker
by David Henningsson 29 Nov '11

29 Nov '11

In kernels 3.0 and 3.1, no automute control was created unless Headphone pins were present. This patch enables automute for the case where there are just Line Out and Speaker pins. BugLink: http://bugs.launchpad.net/bugs/882693 Cc: stable(a)kernel.org (3.0 and 3.1) Tested-by: Marc Legris <marc.legris(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7dc034..c85ce60 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1605,7 +1605,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } - if (spec->automute && cfg->line_out_pins[0] && + if (cfg->line_out_pins[0] && cfg->speaker_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { @@ -1619,6 +1619,10 @@ static void alc_init_auto_hp(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_FRONT_EVENT); spec->detect_line = 1; + if (!spec->automute) { + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + } } spec->automute_lines = spec->detect_line; } -- 1.7.5.4 --------------030009080904060902080809--

1 0

Re: [alsa-devel] MAYA44 USB audio device no inputs control
by Kamil Sobolak 29 Nov '11

29 Nov '11

Hi all! This device has two stereo line ins and we are unable to control them. They are at some unchangable predefined level, but in Windows there are four independent controls of them in Maya44 Mixer. The predefined level of these inputs is so low that we have to use very high gain on microphones and aux signals on our mixing console, which probably isn't good for audio quality. If we used software gain then we would loose audio resolution instead, and also quality would be lower than using appropriate input levels, especially if these are not digital controllers of digital gain, but real controls of preamps that amplify the singal before sending to AD converter. I attached lsusb -v. Thank's for all your help! 2011/11/22 Kamil Sobolak <krzakx(a)gmail.com> > > > Forwarded conversation > Subject: MAYA44 USB audio device no inputs control > ------------------------ > > From: *Kamil Sobolak* <krzakx(a)gmail.com> > Date: 2011/11/22 > To: alsa-devel(a)alsa-project.org > > > Hi! > In our Youth Polish Radio we are using MAYA44 USB sound device, with JACK > and Internet DJ Console (IDJC) to make lives show. > On PC is Ubuntu 11.04 with alsa v1.0.24 and alsamixer but ther is no input > control. > >> ALSA: ice1724 - Fix ESI Maya44 capture source control > > > But this doesn't work. Now after contact with Takashi Iwai I know this > entry is for PCI sound card version not USB. > I have no more idea how to fix it, I have tried many thinks including > booting Ubuntu 11.10 because there is 3.0 kernel but there is still no > input controls. > > I have newest alsa 1.0.24, and can't control input levels. > Here > http://www.mail-archive.com/mixxx-devel@lists.sourceforge.net/msg03598.html somebody > have similar problem, but he uses Maya to vinyl control. > Please help me and for my Youth Polish Radio http://Bieszczady.FM we > would appreciate! > > -- > Kamil Sobolak > > > > ---------- > From: *Daniel Mack* <zonque(a)gmail.com> > Date: 2011/11/22 > To: alsa-devel(a)alsa-project.org > > > Send us your output of 'lsusb -v', please. > > What kind of input control would you expect there to be, and why? > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > -- > Kamil Sobolak > > > -- Kamil Sobolak

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Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Takashi Iwai 29 Nov '11

29 Nov '11

At Mon, 28 Nov 2011 15:19:53 -0600, Pierre-Louis Bossart wrote: > > > A general note: it's not necessarily to split playback and capture > > devices. For example, rawmidi has a single device for full-duplex. > > In the case of PCM, we had to split them because of mmap. The mmap > > for the capture was supposed not to be read-only, some apps may want > > to add mark on the buffer. > > Beats me why an app would want to write anything into the capture ring > buffer. Do you have an example of this use case? Maybe there's a similar > need for compressed data. Thanks! I don't remember exactly, as it was a discussion over 10 years ago. My vague memory says that it's for marking the position in the buffer where the read was finished or so... I think the only possible problem is a case when the device is opened via O_RDWR for full-duplex. When you want mmap in that case, it's difficult to get two different maps for read and write. You may pass different PROT_* flags to distinguish between read and write buffers, but this will really limit the permission. That was the argument. In other words, if the driver doesn't accept O_RDWR but only either O_RDONLY or O_WRONLY, there shouldn't be any problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] [ALSA] usb-audio - Support for Roland GAIA SH-01 Synthesizer
by John F Leach 29 Nov '11

29 Nov '11

Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon Roland SH-201 table entry as template. USB MIDI and audio was tested with Muse and Audacity. Signed-off-by: John F Leach <jfleach(a)jfleach.com> Cc: Daniel Mack <zonque(a)gmail.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 31 +++++++++++++++++++++++++++++++ 1 files changed, 31 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f..6380665 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1400,6 +1400,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Roland GAIA SH-01 */ + USB_DEVICE(0x0582, 0x0111), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "GAIA", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.0.4

3 9

[alsa-devel] [PATCH 1/8] ASoC: Convert atmel directory to module_platform_driver
by Axel Lin 29 Nov '11

29 Nov '11

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/atmel-pcm.c | 12 +----------- sound/soc/atmel/atmel_ssc_dai.c | 12 +----------- 2 files changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index f81d4c3..60de055 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -495,17 +495,7 @@ static struct platform_driver atmel_pcm_driver = { .remove = __devexit_p(atmel_soc_platform_remove), }; -static int __init snd_atmel_pcm_init(void) -{ - return platform_driver_register(&atmel_pcm_driver); -} -module_init(snd_atmel_pcm_init); - -static void __exit snd_atmel_pcm_exit(void) -{ - platform_driver_unregister(&atmel_pcm_driver); -} -module_exit(snd_atmel_pcm_exit); +module_platform_driver(atmel_pcm_driver); MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("Atmel PCM module"); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index a67fc9b..354341e 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -859,17 +859,7 @@ int atmel_ssc_set_audio(int ssc_id) } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -static int __init snd_atmel_ssc_init(void) -{ - return platform_driver_register(&asoc_ssc_driver); -} -module_init(snd_atmel_ssc_init); - -static void __exit snd_atmel_ssc_exit(void) -{ - platform_driver_unregister(&asoc_ssc_driver); -} -module_exit(snd_atmel_ssc_exit); +module_platform_driver(asoc_ssc_driver); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou(a)atmel.com, www.atmel.com"); -- 1.7.5.4

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[alsa-devel] Error from Alsa Lib Test/pcm.c
by ajay khandelwal 29 Nov '11

29 Nov '11

Hi All, I was validating pause/resume of my alsa driver for spdif. After few loop of pause/resume snd_pcm_writei routine returns error. (error code -77, No such device or address). For testing I'm using pcm.c in test/pcm.c. I modified pcm.c under name pcm_pause.c to support pause and resume. I'm attaching my patch for changes below. --- pcm.c 2011-08-13 14:35:15.384157182 +0530 +++ pcm_pause.c 2011-11-30 14:20:26.990180303 +0530 @@ -779,7 +779,7 @@ {NULL, 0, NULL, 0}, }; snd_pcm_t *handle; - int err, morehelp; + int err, morehelp,pid; snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int method = 0; @@ -912,14 +912,40 @@ areas[chn].first = chn * snd_pcm_format_physical_width(format); areas[chn].step = channels * snd_pcm_format_physical_width(format); } - - err = transfer_methods[method].transfer_loop(handle, samples, areas); - if (err < 0) - printf("Transfer failed: %s\n", snd_strerror(err)); - - free(areas); - free(samples); - snd_pcm_close(handle); - return 0; + + pid = fork(); + if(pid > 0){ + err = transfer_methods[method].transfer_loop(handle, samples, areas); + if (err < 0) + printf("Transfer failed: %s\n", snd_strerror(err)); + + free(areas); + free(samples); + snd_pcm_close(handle); + return 0; + } + + if(pid == 0) + { + int pause_count = 0; + int resume_count = 0; + while(1) + { + sleep(1); + if((err = snd_pcm_pause(handle,1)) < 0) + { + printf("\nPause Failed: %s", snd_strerror(err)); + exit(EXIT_FAILURE); + } + printf("\nPAUSED Audio: Count %d\n", ++pause_count); + sleep(1); + if((err = snd_pcm_pause(handle,0)) < 0)\ + { + printf("\nResume Failed: %s", snd_strerror(err)); + exit(EXIT_FAILURE); + } + printf("\nRESUME Audio: Count %d\n", ++resume_count); + } + } }

1 0

[alsa-devel] [PATCH] ASoC: Tegra: Add support of tegra boards based on ALC5632 codec
by Leon Romanovsky 29 Nov '11

29 Nov '11

At this stage only Toshiba AC100/Dynabook supported. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> --- sound/soc/tegra/Kconfig | 9 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_alc5632.c diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c6af1fd..f8fcda3 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -47,3 +47,12 @@ config SND_SOC_TEGRA_TRIMSLICE help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. + +config SND_SOC_TEGRA_ALC5632 + tristate "SoC Audio support for Tegra boards using an ALC5632 codec" + depends on SND_SOC_TEGRA && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_ALC5632 + help + Say Y or M here if you want to add support for SoC audio on the + Toshiba AC100 netbook. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 4d943b3..8e584b8 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -14,6 +14,8 @@ obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-trimslice-objs := trimslice.o +snd-soc-tegra-alc5632-objs := tegra_alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o +obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c new file mode 100644 index 0000000..b8828c1 --- /dev/null +++ b/sound/soc/tegra/tegra_alc5632.c @@ -0,0 +1,298 @@ +/* +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Marc Dietrich <marvin24(a)gmx.de> +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +*/ + +#include <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/alc5632.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-alc5632" + +struct tegra_alc5632 { + struct tegra_asoc_utils_data util_data; +}; + +static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 512 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(&alc5632->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops tegra_alc5632_asoc_ops = { + .hw_params = tegra_alc5632_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_alc5632_hs_jack; + +static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_alc5632_hs_jack_gpios[] = { + { + .name = "Headset Detect", + .report = SND_JACK_HEADSET, + .debounce_time = 150, + } +}; + +static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { + /* Internal Mic */ + {"MIC2", NULL, "Mic Bias 2"}, + {"Mic Bias 2", NULL, "Int Mic"}, + + /* Internal Speaker */ + {"Int Spk", NULL, "SPKOUT"}, + {"Int Spk", NULL, "SPKOUTN"}, + + /* Headset Mic */ + {"MIC1", NULL, "Mic Bias 1"}, + {"Mic Bias 1", NULL, "Headset Mic"}, + + /* Headset Stereophone */ + {"Headset Stereophone", NULL, "HPR"}, + {"Headset Stereophone", NULL, "HPL"}, +}; + +static const struct snd_kcontrol_new tegra_alc5632_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_add_controls(codec, tegra_alc5632_controls, + ARRAY_SIZE(tegra_alc5632_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(dapm, tegra_alc5632_dapm_widgets, + ARRAY_SIZE(tegra_alc5632_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, tegra_alc5632_audio_map, + ARRAY_SIZE(tegra_alc5632_audio_map)); + + snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack); + snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins), + tegra_alc5632_hs_jack_pins); + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + ARRAY_SIZE(tegra_alc5632_hs_jack_gpios), + tegra_alc5632_hs_jack_gpios); + + snd_soc_dapm_nc_pin(dapm, "AUXOUT"); + snd_soc_dapm_nc_pin(dapm, "LINEINL"); + snd_soc_dapm_nc_pin(dapm, "LINEINR"); + snd_soc_dapm_nc_pin(dapm, "PHONEP"); + snd_soc_dapm_nc_pin(dapm, "PHONEN"); + snd_soc_dapm_nc_pin(dapm, "MIC2"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link tegra_alc5632_dai = { + .name = "ALC5632", + .stream_name = "ALC5632 PCM", + .codec_name = "alc5632.0-001e", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "alc5632-hifi", + .init = tegra_alc5632_asoc_init, + .ops = &tegra_alc5632_asoc_ops, +}; + +static struct snd_soc_card snd_soc_tegra_alc5632 = { + .name = "tegra-alc5632", + .dai_link = &tegra_alc5632_dai, + .num_links = 1, +}; + +static __devinit int tegra_alc5632_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_alc5632; + struct tegra_alc5632 *alc5632; + int ret; + + if (!machine_is_paz00()) { + dev_err(&pdev->dev, "Not running on Toshiba AC100!\n"); + return -ENODEV; + } + + alc5632 = kzalloc(sizeof(struct tegra_alc5632), GFP_KERNEL); + if (!alc5632) { + dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + goto err_free_tegra_alc5632; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, alc5632); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_clear_drvdata; + } + + return 0; + +err_clear_drvdata: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + tegra_asoc_utils_fini(&alc5632->util_data); +err_free_tegra_alc5632: + kfree(alc5632); + return ret; +} + +static int __devexit tegra_alc5632_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + + tegra_asoc_utils_fini(&alc5632->util_data); + + kfree(alc5632); + + return 0; +} + +static struct platform_driver tegra_alc5632_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_alc5632_probe, + .remove = __devexit_p(tegra_alc5632_remove), +}; + +static int __init tegra_alc5632_init(void) +{ + return platform_driver_register(&tegra_alc5632_driver); +} +module_init(tegra_alc5632_init); + +static void __exit tegra_alc5632_exit(void) +{ + platform_driver_unregister(&tegra_alc5632_driver); +} +module_exit(tegra_alc5632_exit); + +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- 1.7.3.4

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