From fe81ad1c2d8149323e4a63c5a3bf8b170597c8b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 8 Oct 2012 15:15:46 +0900 Subject: ASoC: wm5102: Write register value corrections after SYSCLK is enabled Evalation of the WM5102 has identified a number of register values which should be written after SYSCLK is enabled on revision A in order to improve performance. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 552 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 551 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1722b586bdba..7394e73fa43c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -42,6 +42,556 @@ static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static const struct reg_default wm5102_sysclk_reva_patch[] = { + { 0x3000, 0x2225 }, + { 0x3001, 0x3a03 }, + { 0x3002, 0x0225 }, + { 0x3003, 0x0801 }, + { 0x3004, 0x6249 }, + { 0x3005, 0x0c04 }, + { 0x3006, 0x0225 }, + { 0x3007, 0x5901 }, + { 0x3008, 0xe249 }, + { 0x3009, 0x030d }, + { 0x300a, 0x0249 }, + { 0x300b, 0x2c01 }, + { 0x300c, 0xe249 }, + { 0x300d, 0x4342 }, + { 0x300e, 0xe249 }, + { 0x300f, 0x73c0 }, + { 0x3010, 0x4249 }, + { 0x3011, 0x0c00 }, + { 0x3012, 0x0225 }, + { 0x3013, 0x1f01 }, + { 0x3014, 0x0225 }, + { 0x3015, 0x1e01 }, + { 0x3016, 0x0225 }, + { 0x3017, 0xfa00 }, + { 0x3018, 0x0000 }, + { 0x3019, 0xf000 }, + { 0x301a, 0x0000 }, + { 0x301b, 0xf000 }, + { 0x301c, 0x0000 }, + { 0x301d, 0xf000 }, + { 0x301e, 0x0000 }, + { 0x301f, 0xf000 }, + { 0x3020, 0x0000 }, + { 0x3021, 0xf000 }, + { 0x3022, 0x0000 }, + { 0x3023, 0xf000 }, + { 0x3024, 0x0000 }, + { 0x3025, 0xf000 }, + { 0x3026, 0x0000 }, + { 0x3027, 0xf000 }, + { 0x3028, 0x0000 }, + { 0x3029, 0xf000 }, + { 0x302a, 0x0000 }, + { 0x302b, 0xf000 }, + { 0x302c, 0x0000 }, + { 0x302d, 0xf000 }, + { 0x302e, 0x0000 }, + { 0x302f, 0xf000 }, + { 0x3030, 0x0225 }, + { 0x3031, 0x1a01 }, + { 0x3032, 0x0225 }, + { 0x3033, 0x1e00 }, + { 0x3034, 0x0225 }, + { 0x3035, 0x1f00 }, + { 0x3036, 0x6225 }, + { 0x3037, 0xf800 }, + { 0x3038, 0x0000 }, + { 0x3039, 0xf000 }, + { 0x303a, 0x0000 }, + { 0x303b, 0xf000 }, + { 0x303c, 0x0000 }, + { 0x303d, 0xf000 }, + { 0x303e, 0x0000 }, + { 0x303f, 0xf000 }, + { 0x3040, 0x2226 }, + { 0x3041, 0x3a03 }, + { 0x3042, 0x0226 }, + { 0x3043, 0x0801 }, + { 0x3044, 0x6249 }, + { 0x3045, 0x0c06 }, + { 0x3046, 0x0226 }, + { 0x3047, 0x5901 }, + { 0x3048, 0xe249 }, + { 0x3049, 0x030d }, + { 0x304a, 0x0249 }, + { 0x304b, 0x2c01 }, + { 0x304c, 0xe249 }, + { 0x304d, 0x4342 }, + { 0x304e, 0xe249 }, + { 0x304f, 0x73c0 }, + { 0x3050, 0x4249 }, + { 0x3051, 0x0c00 }, + { 0x3052, 0x0226 }, + { 0x3053, 0x1f01 }, + { 0x3054, 0x0226 }, + { 0x3055, 0x1e01 }, + { 0x3056, 0x0226 }, + { 0x3057, 0xfa00 }, + { 0x3058, 0x0000 }, + { 0x3059, 0xf000 }, + { 0x305a, 0x0000 }, + { 0x305b, 0xf000 }, + { 0x305c, 0x0000 }, + { 0x305d, 0xf000 }, + { 0x305e, 0x0000 }, + { 0x305f, 0xf000 }, + { 0x3060, 0x0000 }, + { 0x3061, 0xf000 }, + { 0x3062, 0x0000 }, + { 0x3063, 0xf000 }, + { 0x3064, 0x0000 }, + { 0x3065, 0xf000 }, + { 0x3066, 0x0000 }, + { 0x3067, 0xf000 }, + { 0x3068, 0x0000 }, + { 0x3069, 0xf000 }, + { 0x306a, 0x0000 }, + { 0x306b, 0xf000 }, + { 0x306c, 0x0000 }, + { 0x306d, 0xf000 }, + { 0x306e, 0x0000 }, + { 0x306f, 0xf000 }, + { 0x3070, 0x0226 }, + { 0x3071, 0x1a01 }, + { 0x3072, 0x0226 }, + { 0x3073, 0x1e00 }, + { 0x3074, 0x0226 }, + { 0x3075, 0x1f00 }, + { 0x3076, 0x6226 }, + { 0x3077, 0xf800 }, + { 0x3078, 0x0000 }, + { 0x3079, 0xf000 }, + { 0x307a, 0x0000 }, + { 0x307b, 0xf000 }, + { 0x307c, 0x0000 }, + { 0x307d, 0xf000 }, + { 0x307e, 0x0000 }, + { 0x307f, 0xf000 }, + { 0x3080, 0x2227 }, + { 0x3081, 0x3a03 }, + { 0x3082, 0x0227 }, + { 0x3083, 0x0801 }, + { 0x3084, 0x6255 }, + { 0x3085, 0x0c04 }, + { 0x3086, 0x0227 }, + { 0x3087, 0x5901 }, + { 0x3088, 0xe255 }, + { 0x3089, 0x030d }, + { 0x308a, 0x0255 }, + { 0x308b, 0x2c01 }, + { 0x308c, 0xe255 }, + { 0x308d, 0x4342 }, + { 0x308e, 0xe255 }, + { 0x308f, 0x73c0 }, + { 0x3090, 0x4255 }, + { 0x3091, 0x0c00 }, + { 0x3092, 0x0227 }, + { 0x3093, 0x1f01 }, + { 0x3094, 0x0227 }, + { 0x3095, 0x1e01 }, + { 0x3096, 0x0227 }, + { 0x3097, 0xfa00 }, + { 0x3098, 0x0000 }, + { 0x3099, 0xf000 }, + { 0x309a, 0x0000 }, + { 0x309b, 0xf000 }, + { 0x309c, 0x0000 }, + { 0x309d, 0xf000 }, + { 0x309e, 0x0000 }, + { 0x309f, 0xf000 }, + { 0x30a0, 0x0000 }, + { 0x30a1, 0xf000 }, + { 0x30a2, 0x0000 }, + { 0x30a3, 0xf000 }, + { 0x30a4, 0x0000 }, + { 0x30a5, 0xf000 }, + { 0x30a6, 0x0000 }, + { 0x30a7, 0xf000 }, + { 0x30a8, 0x0000 }, + { 0x30a9, 0xf000 }, + { 0x30aa, 0x0000 }, + { 0x30ab, 0xf000 }, + { 0x30ac, 0x0000 }, + { 0x30ad, 0xf000 }, + { 0x30ae, 0x0000 }, + { 0x30af, 0xf000 }, + { 0x30b0, 0x0227 }, + { 0x30b1, 0x1a01 }, + { 0x30b2, 0x0227 }, + { 0x30b3, 0x1e00 }, + { 0x30b4, 0x0227 }, + { 0x30b5, 0x1f00 }, + { 0x30b6, 0x6227 }, + { 0x30b7, 0xf800 }, + { 0x30b8, 0x0000 }, + { 0x30b9, 0xf000 }, + { 0x30ba, 0x0000 }, + { 0x30bb, 0xf000 }, + { 0x30bc, 0x0000 }, + { 0x30bd, 0xf000 }, + { 0x30be, 0x0000 }, + { 0x30bf, 0xf000 }, + { 0x30c0, 0x2228 }, + { 0x30c1, 0x3a03 }, + { 0x30c2, 0x0228 }, + { 0x30c3, 0x0801 }, + { 0x30c4, 0x6255 }, + { 0x30c5, 0x0c06 }, + { 0x30c6, 0x0228 }, + { 0x30c7, 0x5901 }, + { 0x30c8, 0xe255 }, + { 0x30c9, 0x030d }, + { 0x30ca, 0x0255 }, + { 0x30cb, 0x2c01 }, + { 0x30cc, 0xe255 }, + { 0x30cd, 0x4342 }, + { 0x30ce, 0xe255 }, + { 0x30cf, 0x73c0 }, + { 0x30d0, 0x4255 }, + { 0x30d1, 0x0c00 }, + { 0x30d2, 0x0228 }, + { 0x30d3, 0x1f01 }, + { 0x30d4, 0x0228 }, + { 0x30d5, 0x1e01 }, + { 0x30d6, 0x0228 }, + { 0x30d7, 0xfa00 }, + { 0x30d8, 0x0000 }, + { 0x30d9, 0xf000 }, + { 0x30da, 0x0000 }, + { 0x30db, 0xf000 }, + { 0x30dc, 0x0000 }, + { 0x30dd, 0xf000 }, + { 0x30de, 0x0000 }, + { 0x30df, 0xf000 }, + { 0x30e0, 0x0000 }, + { 0x30e1, 0xf000 }, + { 0x30e2, 0x0000 }, + { 0x30e3, 0xf000 }, + { 0x30e4, 0x0000 }, + { 0x30e5, 0xf000 }, + { 0x30e6, 0x0000 }, + { 0x30e7, 0xf000 }, + { 0x30e8, 0x0000 }, + { 0x30e9, 0xf000 }, + { 0x30ea, 0x0000 }, + { 0x30eb, 0xf000 }, + { 0x30ec, 0x0000 }, + { 0x30ed, 0xf000 }, + { 0x30ee, 0x0000 }, + { 0x30ef, 0xf000 }, + { 0x30f0, 0x0228 }, + { 0x30f1, 0x1a01 }, + { 0x30f2, 0x0228 }, + { 0x30f3, 0x1e00 }, + { 0x30f4, 0x0228 }, + { 0x30f5, 0x1f00 }, + { 0x30f6, 0x6228 }, + { 0x30f7, 0xf800 }, + { 0x30f8, 0x0000 }, + { 0x30f9, 0xf000 }, + { 0x30fa, 0x0000 }, + { 0x30fb, 0xf000 }, + { 0x30fc, 0x0000 }, + { 0x30fd, 0xf000 }, + { 0x30fe, 0x0000 }, + { 0x30ff, 0xf000 }, + { 0x3100, 0x222b }, + { 0x3101, 0x3a03 }, + { 0x3102, 0x222b }, + { 0x3103, 0x5803 }, + { 0x3104, 0xe26f }, + { 0x3105, 0x030d }, + { 0x3106, 0x626f }, + { 0x3107, 0x2c01 }, + { 0x3108, 0xe26f }, + { 0x3109, 0x4342 }, + { 0x310a, 0xe26f }, + { 0x310b, 0x73c0 }, + { 0x310c, 0x026f }, + { 0x310d, 0x0c00 }, + { 0x310e, 0x022b }, + { 0x310f, 0x1f01 }, + { 0x3110, 0x022b }, + { 0x3111, 0x1e01 }, + { 0x3112, 0x022b }, + { 0x3113, 0xfa00 }, + { 0x3114, 0x0000 }, + { 0x3115, 0xf000 }, + { 0x3116, 0x0000 }, + { 0x3117, 0xf000 }, + { 0x3118, 0x0000 }, + { 0x3119, 0xf000 }, + { 0x311a, 0x0000 }, + { 0x311b, 0xf000 }, + { 0x311c, 0x0000 }, + { 0x311d, 0xf000 }, + { 0x311e, 0x0000 }, + { 0x311f, 0xf000 }, + { 0x3120, 0x022b }, + { 0x3121, 0x0a01 }, + { 0x3122, 0x022b }, + { 0x3123, 0x1e00 }, + { 0x3124, 0x022b }, + { 0x3125, 0x1f00 }, + { 0x3126, 0x622b }, + { 0x3127, 0xf800 }, + { 0x3128, 0x0000 }, + { 0x3129, 0xf000 }, + { 0x312a, 0x0000 }, + { 0x312b, 0xf000 }, + { 0x312c, 0x0000 }, + { 0x312d, 0xf000 }, + { 0x312e, 0x0000 }, + { 0x312f, 0xf000 }, + { 0x3130, 0x0000 }, + { 0x3131, 0xf000 }, + { 0x3132, 0x0000 }, + { 0x3133, 0xf000 }, + { 0x3134, 0x0000 }, + { 0x3135, 0xf000 }, + { 0x3136, 0x0000 }, + { 0x3137, 0xf000 }, + { 0x3138, 0x0000 }, + { 0x3139, 0xf000 }, + { 0x313a, 0x0000 }, + { 0x313b, 0xf000 }, + { 0x313c, 0x0000 }, + { 0x313d, 0xf000 }, + { 0x313e, 0x0000 }, + { 0x313f, 0xf000 }, + { 0x3140, 0x0000 }, + { 0x3141, 0xf000 }, + { 0x3142, 0x0000 }, + { 0x3143, 0xf000 }, + { 0x3144, 0x0000 }, + { 0x3145, 0xf000 }, + { 0x3146, 0x0000 }, + { 0x3147, 0xf000 }, + { 0x3148, 0x0000 }, + { 0x3149, 0xf000 }, + { 0x314a, 0x0000 }, + { 0x314b, 0xf000 }, + { 0x314c, 0x0000 }, + { 0x314d, 0xf000 }, + { 0x314e, 0x0000 }, + { 0x314f, 0xf000 }, + { 0x3150, 0x0000 }, + { 0x3151, 0xf000 }, + { 0x3152, 0x0000 }, + { 0x3153, 0xf000 }, + { 0x3154, 0x0000 }, + { 0x3155, 0xf000 }, + { 0x3156, 0x0000 }, + { 0x3157, 0xf000 }, + { 0x3158, 0x0000 }, + { 0x3159, 0xf000 }, + { 0x315a, 0x0000 }, + { 0x315b, 0xf000 }, + { 0x315c, 0x0000 }, + { 0x315d, 0xf000 }, + { 0x315e, 0x0000 }, + { 0x315f, 0xf000 }, + { 0x3160, 0x0000 }, + { 0x3161, 0xf000 }, + { 0x3162, 0x0000 }, + { 0x3163, 0xf000 }, + { 0x3164, 0x0000 }, + { 0x3165, 0xf000 }, + { 0x3166, 0x0000 }, + { 0x3167, 0xf000 }, + { 0x3168, 0x0000 }, + { 0x3169, 0xf000 }, + { 0x316a, 0x0000 }, + { 0x316b, 0xf000 }, + { 0x316c, 0x0000 }, + { 0x316d, 0xf000 }, + { 0x316e, 0x0000 }, + { 0x316f, 0xf000 }, + { 0x3170, 0x0000 }, + { 0x3171, 0xf000 }, + { 0x3172, 0x0000 }, + { 0x3173, 0xf000 }, + { 0x3174, 0x0000 }, + { 0x3175, 0xf000 }, + { 0x3176, 0x0000 }, + { 0x3177, 0xf000 }, + { 0x3178, 0x0000 }, + { 0x3179, 0xf000 }, + { 0x317a, 0x0000 }, + { 0x317b, 0xf000 }, + { 0x317c, 0x0000 }, + { 0x317d, 0xf000 }, + { 0x317e, 0x0000 }, + { 0x317f, 0xf000 }, + { 0x3180, 0x2001 }, + { 0x3181, 0xf101 }, + { 0x3182, 0x0000 }, + { 0x3183, 0xf000 }, + { 0x3184, 0x0000 }, + { 0x3185, 0xf000 }, + { 0x3186, 0x0000 }, + { 0x3187, 0xf000 }, + { 0x3188, 0x0000 }, + { 0x3189, 0xf000 }, + { 0x318a, 0x0000 }, + { 0x318b, 0xf000 }, + { 0x318c, 0x0000 }, + { 0x318d, 0xf000 }, + { 0x318e, 0x0000 }, + { 0x318f, 0xf000 }, + { 0x3190, 0x0000 }, + { 0x3191, 0xf000 }, + { 0x3192, 0x0000 }, + { 0x3193, 0xf000 }, + { 0x3194, 0x0000 }, + { 0x3195, 0xf000 }, + { 0x3196, 0x0000 }, + { 0x3197, 0xf000 }, + { 0x3198, 0x0000 }, + { 0x3199, 0xf000 }, + { 0x319a, 0x0000 }, + { 0x319b, 0xf000 }, + { 0x319c, 0x0000 }, + { 0x319d, 0xf000 }, + { 0x319e, 0x0000 }, + { 0x319f, 0xf000 }, + { 0x31a0, 0x0000 }, + { 0x31a1, 0xf000 }, + { 0x31a2, 0x0000 }, + { 0x31a3, 0xf000 }, + { 0x31a4, 0x0000 }, + { 0x31a5, 0xf000 }, + { 0x31a6, 0x0000 }, + { 0x31a7, 0xf000 }, + { 0x31a8, 0x0000 }, + { 0x31a9, 0xf000 }, + { 0x31aa, 0x0000 }, + { 0x31ab, 0xf000 }, + { 0x31ac, 0x0000 }, + { 0x31ad, 0xf000 }, + { 0x31ae, 0x0000 }, + { 0x31af, 0xf000 }, + { 0x31b0, 0x0000 }, + { 0x31b1, 0xf000 }, + { 0x31b2, 0x0000 }, + { 0x31b3, 0xf000 }, + { 0x31b4, 0x0000 }, + { 0x31b5, 0xf000 }, + { 0x31b6, 0x0000 }, + { 0x31b7, 0xf000 }, + { 0x31b8, 0x0000 }, + { 0x31b9, 0xf000 }, + { 0x31ba, 0x0000 }, + { 0x31bb, 0xf000 }, + { 0x31bc, 0x0000 }, + { 0x31bd, 0xf000 }, + { 0x31be, 0x0000 }, + { 0x31bf, 0xf000 }, + { 0x31c0, 0x0000 }, + { 0x31c1, 0xf000 }, + { 0x31c2, 0x0000 }, + { 0x31c3, 0xf000 }, + { 0x31c4, 0x0000 }, + { 0x31c5, 0xf000 }, + { 0x31c6, 0x0000 }, + { 0x31c7, 0xf000 }, + { 0x31c8, 0x0000 }, + { 0x31c9, 0xf000 }, + { 0x31ca, 0x0000 }, + { 0x31cb, 0xf000 }, + { 0x31cc, 0x0000 }, + { 0x31cd, 0xf000 }, + { 0x31ce, 0x0000 }, + { 0x31cf, 0xf000 }, + { 0x31d0, 0x0000 }, + { 0x31d1, 0xf000 }, + { 0x31d2, 0x0000 }, + { 0x31d3, 0xf000 }, + { 0x31d4, 0x0000 }, + { 0x31d5, 0xf000 }, + { 0x31d6, 0x0000 }, + { 0x31d7, 0xf000 }, + { 0x31d8, 0x0000 }, + { 0x31d9, 0xf000 }, + { 0x31da, 0x0000 }, + { 0x31db, 0xf000 }, + { 0x31dc, 0x0000 }, + { 0x31dd, 0xf000 }, + { 0x31de, 0x0000 }, + { 0x31df, 0xf000 }, + { 0x31e0, 0x0000 }, + { 0x31e1, 0xf000 }, + { 0x31e2, 0x0000 }, + { 0x31e3, 0xf000 }, + { 0x31e4, 0x0000 }, + { 0x31e5, 0xf000 }, + { 0x31e6, 0x0000 }, + { 0x31e7, 0xf000 }, + { 0x31e8, 0x0000 }, + { 0x31e9, 0xf000 }, + { 0x31ea, 0x0000 }, + { 0x31eb, 0xf000 }, + { 0x31ec, 0x0000 }, + { 0x31ed, 0xf000 }, + { 0x31ee, 0x0000 }, + { 0x31ef, 0xf000 }, + { 0x31f0, 0x0000 }, + { 0x31f1, 0xf000 }, + { 0x31f2, 0x0000 }, + { 0x31f3, 0xf000 }, + { 0x31f4, 0x0000 }, + { 0x31f5, 0xf000 }, + { 0x31f6, 0x0000 }, + { 0x31f7, 0xf000 }, + { 0x31f8, 0x0000 }, + { 0x31f9, 0xf000 }, + { 0x31fa, 0x0000 }, + { 0x31fb, 0xf000 }, + { 0x31fc, 0x0000 }, + { 0x31fd, 0xf000 }, + { 0x31fe, 0x0000 }, + { 0x31ff, 0xf000 }, + { 0x024d, 0xff50 }, + { 0x0252, 0xff50 }, + { 0x0259, 0x0112 }, + { 0x025e, 0x0112 }, +}; + +static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm5102_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm5102_sysclk_reva_patch); + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static const struct snd_kcontrol_new wm5102_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, 1, 0), @@ -297,7 +847,7 @@ static const struct snd_kcontrol_new wm5102_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5102_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, -- cgit v1.2.3 From 9f4c3f1cde541d477633479a0203ef8a834ee5f9 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 31 Oct 2012 01:20:05 -0200 Subject: ASoC: mxs-saif: Add MODULE_ALIAS Add MODULE_ALIAS information. Signed-off-by: Fabio Estevam Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index aa037b292f3d..93380cc7cf97 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -812,3 +812,4 @@ module_platform_driver(mxs_saif_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("MXS ASoC SAIF driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mxs-saif"); -- cgit v1.2.3 From 213a79656462176b553c6f9cdf96e14313e43bcf Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 2 Nov 2012 13:02:53 +0000 Subject: ASoC: bells: Add missing select of WM0010 Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e7b83179aca2..fa166bd87edc 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -207,6 +207,7 @@ config SND_SOC_BELLS select SND_SOC_WM5102 select SND_SOC_WM5110 select SND_SOC_WM9081 + select SND_SOC_WM0010 config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" -- cgit v1.2.3 From 4868ce57bfe1810262231dd8fe83fbba0ab59f13 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 2 Nov 2012 13:02:54 +0000 Subject: ASoC: bells: Select WM1250-EV1 Springbank audio I/O module Ensure we select the WM1250-EV1 as the current software system configuration demands it. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fa166bd87edc..3c7c3a59ed39 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -208,6 +208,7 @@ config SND_SOC_BELLS select SND_SOC_WM5110 select SND_SOC_WM9081 select SND_SOC_WM0010 + select SND_SOC_WM1250_EV1 config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" -- cgit v1.2.3 From f55f14752ecaccf7d6a52fd13929b73fcb191f19 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 1 Nov 2012 15:57:11 -0200 Subject: ASoC: mxs-saif: Fix channel swap for 24-bit format Playing 24-bit format file leads to channel swap on mx28 and the reason is that the current driver performs one write/read to/from the SAIF_DATA register to trigger the transfer. This approach works fine for S16_LE case because SAIF_DATA is a 32-bit register and thus is capable of storing the 16-bit left and right channels, but for the S24_LE case it can only store one channel, so in order to not lose the FIFO sync an extra read/write is needed. Reported-by: Dan Winner Signed-off-by: Fabio Estevam Tested-by: Dan Winner Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 93380cc7cf97..c294fbb523fc 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -523,16 +523,24 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* - * write a data to saif data register to trigger - * the transfer + * write data to saif data register to trigger + * the transfer. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to write twice. + * This is also safe for the other non 24-bit formats. */ __raw_writel(0, saif->base + SAIF_DATA); + __raw_writel(0, saif->base + SAIF_DATA); } else { /* - * read a data from saif data register to trigger - * the receive + * read data from saif data register to trigger + * the receive. + * For 24-bit format the 32-bit FIFO register stores + * only one channel, so we need to read twice. + * This is also safe for the other non 24-bit formats. */ __raw_readl(saif->base + SAIF_DATA); + __raw_readl(saif->base + SAIF_DATA); } master_saif->ongoing = 1; -- cgit v1.2.3 From ec8f53fb693dda095ad3342b927a074e7c4dddfa Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 2 Nov 2012 00:28:50 +0900 Subject: ALSA: Fix typo in drivers sound Correct spelling typo in debug messages within drivers/sound Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4113.c | 2 +- sound/i2c/other/ak4114.c | 2 +- sound/i2c/other/ak4117.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index ef68d710d08c..e04e750a77ed 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -426,7 +426,7 @@ static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4113_spdif_pinfo, diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index 816e7d225fb0..5bf4fca19e48 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -401,7 +401,7 @@ static struct snd_kcontrol_new snd_ak4114_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4114_spdif_pinfo, .get = snd_ak4114_spdif_pget, diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index b4b2a51fc117..40e33c9f2b09 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -380,7 +380,7 @@ static struct snd_kcontrol_new snd_ak4117_iec958_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = "IEC958 Preample Capture Default", + .name = "IEC958 Preamble Capture Default", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ak4117_spdif_pinfo, .get = snd_ak4117_spdif_pget, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f1cd1e387801..9a8d5cef32c7 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4899,7 +4899,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, insel = "Coaxial"; break; default: - insel = "Unkown"; + insel = "Unknown"; } snd_iprintf(buffer, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 61599298fb26..4d8db3685e96 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai, if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { cs42l52->sysclk = freq; } else { - dev_err(codec->dev, "Invalid freq paramter\n"); + dev_err(codec->dev, "Invalid freq parameter\n"); return -EINVAL; } return 0; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3fddc7ad1127..b2b2b37131bd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) } while (count--); if (count == 0) - dev_warn(codec->dev, "No impedence range reported for jack\n"); + dev_warn(codec->dev, "No impedance range reported for jack\n"); #ifndef CONFIG_SND_SOC_WM8994_MODULE trace_snd_soc_jack_irq(dev_name(codec->dev)); -- cgit v1.2.3 From f0b3da98434589a5665d70041f8e1a5600b84fe8 Mon Sep 17 00:00:00 2001 From: "Lars R. Damerow" Date: Fri, 2 Nov 2012 13:10:39 -0700 Subject: ALSA: hda - support Teradici 2200 host card audio The audio chipset used in Teradici's Tera2 host cards is the same as that in the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards. Signed-off-by: Lars R. Damerow Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085ae7d46..cd2dbaf1be78 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3563,6 +3563,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, + { PCI_DEVICE(0x6549, 0x2200), + .driver_data = AZX_DRIVER_TERA | AZX_DCAPS_NO_64BIT }, /* Creative X-Fi (CA0110-IBG) */ /* CTHDA chips */ { PCI_DEVICE(0x1102, 0x0010), -- cgit v1.2.3 From 5a83b4b5a391f07141b157ac9daa51c409e71ab5 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Thu, 1 Nov 2012 13:42:37 +0100 Subject: ALSA: hda: Cirrus: Fix coefficient index for beep configuration Signed-off-by: Alexander Stein Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 61a71131711c..3b7d67af1441 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1107,7 +1107,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { | 0x0400 /* Disable Coefficient Auto increment */ )}, /* Beep */ - {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_BEEP_CFG}, {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ {} /* terminator */ -- cgit v1.2.3 From 16337e028a6dae9fbdd718c0d42161540a668ff3 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:03 +0800 Subject: ALSA: HDA: Fix digital microphone on CS420x Correctly enable the digital microphones with the right bits in the right coeffecient registers on Cirrus CS4206/7 codecs. It also prevents misconfiguring ADC1/2. This fixes the digital mic on the Macbook Pro 10,1/Retina. Based-on-patch-by: Alexander Stein Signed-off-by: Daniel J Blueman Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 3b7d67af1441..859a1197e080 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -101,8 +101,8 @@ enum { #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 #define CS_DIG_OUT2_PIN_NID 0x15 -#define CS_DMIC1_PIN_NID 0x12 -#define CS_DMIC2_PIN_NID 0x0e +#define CS_DMIC1_PIN_NID 0x0e +#define CS_DMIC2_PIN_NID 0x12 /* coef indices */ #define IDX_SPDIF_STAT 0x0000 @@ -1079,14 +1079,18 @@ static void init_input(struct hda_codec *codec) cs_automic(codec, NULL); coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + coef = cs_vendor_coef_get(codec, IDX_BEEP_CFG); if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + coef |= 1 << 4; /* DMIC2 2 chan on, GPIO1 off */ if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + coef |= 1 << 3; /* DMIC1 2 chan on, GPIO0 off * No effect if SPDIF_OUT2 is * selected in IDX_SPDIF_CTL. */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + + cs_vendor_coef_set(codec, IDX_BEEP_CFG, coef); } else { if (spec->mic_detect) cs_automic(codec, NULL); -- cgit v1.2.3 From 00e17f767e3e8d42b83a12af3ed16e3129e4feb0 Mon Sep 17 00:00:00 2001 From: Daniel J Blueman Date: Sun, 4 Nov 2012 13:19:04 +0800 Subject: ALSA: HDA: Mark CS260x immutable structures const Mark structures that won't change const. Signed-off-by: Daniel J Blueman Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 859a1197e080..d5f3a26d608d 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1732,8 +1732,7 @@ static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new cs421x_capture_source = { - +static const struct snd_kcontrol_new cs421x_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -1950,7 +1949,7 @@ static int cs421x_suspend(struct hda_codec *codec) } #endif -static struct hda_codec_ops cs421x_patch_ops = { +static const struct hda_codec_ops cs421x_patch_ops = { .build_controls = cs421x_build_controls, .build_pcms = cs_build_pcms, .init = cs421x_init, -- cgit v1.2.3 From 5c0ee9497b33cde3e57460efe4f73313dc0b57a3 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sun, 4 Nov 2012 23:34:58 +0100 Subject: ALSA: es1968: Add ESS vendor ID to pm_whitelist Add generic ESS vendor ID to pm_whitelist. This should fix suspend on all Maestro-2 and Maestro-2E based PCI cards. Tested on Terratec DMX and SF64-PCE2. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 5d0e568fdea1..50169bcfd903 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2655,6 +2655,8 @@ static struct ess_device_list pm_whitelist[] __devinitdata = { { TYPE_MAESTRO2E, 0x1179 }, { TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */ { TYPE_MAESTRO2E, 0x1558 }, + { TYPE_MAESTRO2E, 0x125d }, /* a PCI card, e.g. Terratec DMX */ + { TYPE_MAESTRO2, 0x125d }, /* a PCI card, e.g. SF64-PCE2 */ }; static struct ess_device_list mpu_blacklist[] __devinitdata = { -- cgit v1.2.3 From ae24c3191ba2ab03ec6b4be323e730e00404b4b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Nov 2012 12:32:46 +0100 Subject: ALSA: hda - Force to reset IEC958 status bits for AD codecs Several bug reports suggest that the forcibly resetting IEC958 status bits is required for AD codecs to get the SPDIF output working properly after changing streams. Original fix credit to Javeed Shaikh. BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361 Reported-by: Robin Kreis Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdd43eadbc67..1eeba7386666 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -545,6 +545,7 @@ static int ad198x_build_pcms(struct hda_codec *codec) if (spec->multiout.dig_out_nid) { info++; codec->num_pcms++; + codec->spdif_status_reset = 1; info->name = "AD198x Digital"; info->pcm_type = HDA_PCM_TYPE_SPDIF; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; -- cgit v1.2.3 From 55c6f4cb6ef49afbb86222c6a3ff85329199c729 Mon Sep 17 00:00:00 2001 From: Eric Millbrandt Date: Fri, 2 Nov 2012 17:05:44 -0400 Subject: ASoC: wm8978: pll incorrectly configured when codec is master When MCLK is supplied externally and BCLK and LRC are configured as outputs (codec is master), the PLL values are only calculated correctly on the first transmission. On subsequent transmissions, at differenct sample rates, the wrong PLL values are used. Test for f_opclk instead of f_pllout to determine if the PLL values are needed. Signed-off-by: Eric Millbrandt Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8978.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 5421fd9fbcb5..4c0a8e496131 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -782,7 +782,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, wm8978->mclk_idx = -1; f_sel = wm8978->f_mclk; } else { - if (!wm8978->f_pllout) { + if (!wm8978->f_opclk) { /* We only enter here, if OPCLK is not used */ int ret = wm8978_configure_pll(codec); if (ret < 0) -- cgit v1.2.3 From 5b3761954dac2d1393beef8210eb8cee81d16b8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:32:47 +0100 Subject: ALSA: hda - Fix empty DAC filling in patch_via.c In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at the point of the current line-out (i). When no valid path is found for this output, this results in dac = 0, thus it creates a hole in dac_nids[]. This confuses is_empty_dac() and trims the detected DAC in later reference. This patch fixes the bug by appending DAC properly to dac_nids[] in via_auto_fill_adc_nids(). Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 72a2f60b087c..bf57fa6a4add 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1809,11 +1809,11 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; - int i, dac_num; + int i; hda_nid_t nid; + spec->multiout.num_dacs = 0; spec->multiout.dac_nids = spec->private_dac_nids; - dac_num = 0; for (i = 0; i < cfg->line_outs; i++) { hda_nid_t dac = 0; nid = cfg->line_out_pins[i]; @@ -1824,16 +1824,13 @@ static int via_auto_fill_dac_nids(struct hda_codec *codec) if (!i && parse_output_path(codec, nid, dac, 1, &spec->out_mix_path)) dac = spec->out_mix_path.path[0]; - if (dac) { - spec->private_dac_nids[i] = dac; - dac_num++; - } + if (dac) + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } if (!spec->out_path[0].depth && spec->out_mix_path.depth) { spec->out_path[0] = spec->out_mix_path; spec->out_mix_path.depth = 0; } - spec->multiout.num_dacs = dac_num; return 0; } -- cgit v1.2.3 From ef4da45828603df57e5e21b8aa21a66ce309f79b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:37:48 +0100 Subject: ALSA: hda - Fix invalid connections in VT1802 codec VT1802 codec provides the invalid connection lists of NID 0x24 and 0x33 containing the routes to a non-exist widget 0x3e. This confuses the auto-parser. Fix it up in the driver by overriding these connections. Reported-by: Massimo Del Fedele Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index bf57fa6a4add..c2eef5cb78d8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3646,6 +3646,18 @@ static const struct snd_pci_quirk vt2002p_fixups[] = { {} }; +/* NIDs 0x24 and 0x33 on VT1802 have connections to non-existing NID 0x3e + * Replace this with mixer NID 0x1c + */ +static void fix_vt1802_connections(struct hda_codec *codec) +{ + static hda_nid_t conn_24[] = { 0x14, 0x1c }; + static hda_nid_t conn_33[] = { 0x1c }; + + snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24); + snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33); +} + /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) { @@ -3660,6 +3672,8 @@ static int patch_vt2002P(struct hda_codec *codec) spec->aa_mix_nid = 0x21; override_mic_boost(codec, 0x2b, 0, 3, 40); override_mic_boost(codec, 0x29, 0, 3, 40); + if (spec->codec_type == VT1802) + fix_vt1802_connections(codec); add_secret_dac_path(codec); snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); -- cgit v1.2.3 From d5266125fb439a5dfa4edd548d888fda47414ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Nov 2012 10:40:36 +0100 Subject: ALSA: hda - Add pin fixups for ASUS G75 To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802 codec, correct the default configurations of speaker pins 0x24 and 0x33. Reported-by: Massimo Del Fedele Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c2eef5cb78d8..019e1a00414a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3625,6 +3625,7 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) */ enum { VIA_FIXUP_INTMIC_BOOST, + VIA_FIXUP_ASUS_G75, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -3639,9 +3640,19 @@ static const struct hda_fixup via_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = via_fixup_intmic_boost, }, + [VIA_FIXUP_ASUS_G75] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* set 0x24 and 0x33 as speakers */ + { 0x24, 0x991301f0 }, + { 0x33, 0x991301f1 }, /* subwoofer */ + { } + } + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), {} }; -- cgit v1.2.3 From 6268f74990c7fab6727bcb2dc82b3c4d4b302317 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 6 Nov 2012 16:33:18 +0000 Subject: ASoC: bells: Correct type in sub speaker DAI name for WM5102 Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index b0d46d63d55e..b56b9a3c6169 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5110-aif3", + .cpu_dai_name = "wm5102-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.2.3 From 5c855c8e2be67f2d5a989ef1190098f924f9f820 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Wed, 7 Nov 2012 20:38:35 +0800 Subject: ASoC: cs42l52: fix the return value of cs42l52_set_fmt() Fix the return value of cs42l52_set_fmt() when clock inversion is not allowed and also remove the useless variable ret. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) [We had been assigning to ret but then ignoring the value we assgined -- broonie] Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l52.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 61599298fb26..f91136caa4c7 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -773,7 +773,6 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); - int ret = 0; u8 iface = 0; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -822,7 +821,7 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_IF: break; default: - ret = -EINVAL; + return -EINVAL; } cs42l52->config.format = iface; snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); -- cgit v1.2.3 From d1a3c98d50731c627909029bb653a0557946f0f5 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Wed, 7 Nov 2012 18:00:09 +0100 Subject: ALSA: hdspm - Fix sync check reporting on RME RayDAT The RayDAT reports the sync status of its inputs in consecutive bit positions, so all we do in hdspm_s1_sync_check is to iterate over idx: status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); lock = (status & (0x1<private_value: HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("AES SyncCheck", 1), HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2), HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3), HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4), HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5), HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6), HDSPM_SYNC_CHECK("TCO SyncCheck", 7), HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8), The patch corrects the indicated sync flags by passing the proper index value to hdspm_s1_sync_check(). Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9a8d5cef32c7..748e36c66603 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3979,7 +3979,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 8: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; -- cgit v1.2.3 From f58161ba1b05a968e5136824b5a16b714b6a5317 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 08:52:45 +0100 Subject: ALSA: usb-audio: Fix crash at re-preparing the PCM stream There are bug reports of a crash with USB-audio devices when PCM prepare is performed immediately after the stream is stopped via trigger callback. It turned out that the problem is that we don't wait until all URBs are killed. This patch adds a new function to synchronize the pending stop operation on an endpoint, and calls in the prepare callback for avoiding the crash above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181 Reported-and-tested-by: Artem S. Tashkinov Cc: [v3.6] Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 13 +++++++++++++ sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 3 +++ 3 files changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7f78c6d782b0..34de6f2faf61 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,6 +35,7 @@ #define EP_FLAG_ACTIVATED 0 #define EP_FLAG_RUNNING 1 +#define EP_FLAG_STOPPING 2 /* * snd_usb_endpoint is a model that abstracts everything related to an @@ -502,10 +503,20 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) if (alive) snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n", alive, ep->ep_num); + clear_bit(EP_FLAG_STOPPING, &ep->flags); return 0; } +/* sync the pending stop operation; + * this function itself doesn't trigger the stop operation + */ +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep) +{ + if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags)) + wait_clear_urbs(ep); +} + /* * unlink active urbs. */ @@ -918,6 +929,8 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, if (wait) wait_clear_urbs(ep); + else + set_bit(EP_FLAG_STOPPING, &ep->flags); } } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6376ccf10fd4..3d4c9705041f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -19,6 +19,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); +void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 37428f74dbb6..5c12a3fe8c3e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -568,6 +568,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint); + snd_usb_endpoint_sync_pending_stop(subs->data_endpoint); + ret = set_format(subs, subs->cur_audiofmt); if (ret < 0) goto unlock; -- cgit v1.2.3 From 1387e2d12799e554df2f60e7ae7fe01384bcb96f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:23:18 +0100 Subject: ALSA: hda - Improve HP depop when system enter to S3 alc269_toggle_power_output() was only use in ALC269VB. I rename it to alc269vb_toggle_power_output(). Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++------------- 1 file changed, 11 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7397ad02a0d..b25e9b22cd69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5840,7 +5840,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } -static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) +static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); if (power_up) @@ -5857,10 +5857,10 @@ static void alc269_shutup(struct hda_codec *codec) if (spec->codec_variant != ALC269_TYPE_ALC269VB) return; - if ((alc_get_coef0(codec) & 0x00ff) == 0x017) - alc269_toggle_power_output(codec, 0); - if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && + (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } } @@ -5870,24 +5870,22 @@ static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 0); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { - alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); - if (spec->codec_variant == ALC269_TYPE_ALC269VB || + if (spec->codec_variant == ALC269_TYPE_ALC269VB) + alc269vb_toggle_power_output(codec, 1); + if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x017) { - alc269_toggle_power_output(codec, 1); msleep(200); } - if (spec->codec_variant == ALC269_TYPE_ALC269VB || - (alc_get_coef0(codec) & 0x00ff) == 0x018) - alc269_toggle_power_output(codec, 1); - snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); hda_call_check_power_status(codec, 0x01); -- cgit v1.2.3 From 19a62823eae453619604636082085812c14ee391 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 8 Nov 2012 10:25:37 +0100 Subject: ALSA: hda - Add new codec ALC668 and ALC900 (default name ALC1150) Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b25e9b22cd69..c0ce3b1f04b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7077,6 +7077,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, + { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, @@ -7094,6 +7095,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 }, + { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 }, {} /* terminator */ }; -- cgit v1.2.3 From 8bb4d9ce08b0a92ca174e41d92c180328f86173f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Nov 2012 14:36:18 +0100 Subject: ALSA: Fix card refcount unbalance There are uncovered cases whether the card refcount introduced by the commit a0830dbd isn't properly increased or decreased: - OSS PCM and mixer success paths - When lookup function gets NULL This patch fixes these places. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251 Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 1 + sound/core/oss/pcm_oss.c | 1 + sound/core/pcm_native.c | 6 ++++-- sound/core/sound.c | 2 +- sound/core/sound_oss.c | 2 +- 5 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index a9a2e63c0222..e8a1d18774b2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -76,6 +76,7 @@ static int snd_mixer_oss_open(struct inode *inode, struct file *file) snd_card_unref(card); return -EFAULT; } + snd_card_unref(card); return 0; } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f337b66a020b..4c1cc51772e6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2454,6 +2454,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; + snd_card_unref(pcm->card); return err; __error: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6e8872de5ba0..f9ddecf2f4cd 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2122,7 +2122,8 @@ static int snd_pcm_playback_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_PLAYBACK); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_PLAYBACK); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } @@ -2135,7 +2136,8 @@ static int snd_pcm_capture_open(struct inode *inode, struct file *file) pcm = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_PCM_CAPTURE); err = snd_pcm_open(file, pcm, SNDRV_PCM_STREAM_CAPTURE); - snd_card_unref(pcm->card); + if (pcm) + snd_card_unref(pcm->card); return err; } diff --git a/sound/core/sound.c b/sound/core/sound.c index 89780c323f19..70ccdab74153 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -114,7 +114,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) mreg = snd_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index e1d79ee35906..726a49ac9725 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -54,7 +54,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) { private_data = mreg->private_data; - if (mreg->card_ptr) + if (private_data && mreg->card_ptr) atomic_inc(&mreg->card_ptr->refcount); } else private_data = NULL; -- cgit v1.2.3 From 445632ad6dda42f4d3f9df2569a852ca0d4ea608 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Thu, 8 Nov 2012 12:03:12 -0600 Subject: ASoC: dapm: Use card_list during DAPM shutdown DAPM shutdown incorrectly uses "list" field of codec struct while iterating over probed components (codec_dev_list). "list" field refers to codecs registered in the system, "card_list" field is used for probed components. Signed-off-by: Misael Lopez Cruz Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0a4be38dc0f..6e35bcae02df 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3745,7 +3745,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { struct snd_soc_codec *codec; - list_for_each_entry(codec, &card->codec_dev_list, list) { + list_for_each_entry(codec, &card->codec_dev_list, card_list) { soc_dapm_shutdown_codec(&codec->dapm); if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) snd_soc_dapm_set_bias_level(&codec->dapm, -- cgit v1.2.3 From d055852ee86703d48b0c571e94bd2eb33aa9b91d Mon Sep 17 00:00:00 2001 From: Mukund Navada Date: Fri, 9 Nov 2012 11:53:40 +0530 Subject: ASoC: core: Double control update err for snd_soc_put_volsw_sx snd_soc_put_volsw_sx function fails to update second control if first control is updated by snd_soc_update_bits_locked. Signed-off-by: Mukund Navada Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d1198627fc40..10d21be383f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2786,8 +2786,9 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0] + min) & mask; val = val << shift; - if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) - return err; + err = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (err < 0) + return err; if (snd_soc_volsw_is_stereo(mc)) { val_mask = mask << rshift; -- cgit v1.2.3 From 05193639ca977cc889668718adb38db6d585045b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Nov 2012 10:07:36 +0100 Subject: ALSA: hda - Add a missing quirk entry for iMac 9,1 This is another variant of iMac 9,1 with a different codec SSID. Reported-and-tested-by: Everaldo Canuto Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c0ce3b1f04b4..68fd49294b26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5407,6 +5407,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), -- cgit v1.2.3 From 5574f7745436d2014fcba1163f820d132e816c85 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 10 Nov 2012 19:52:50 +0100 Subject: ASoC: cs4271: free allocated GPIO In case of probe deferral, the allocated GPIO line is not freed, which prevents it from being claimed and properly asserted in later attempts. Fix this by using devm_gpio_request(). Signed-off-by: Daniel Mack Reported-by: Michael Hirsch Cc: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f994af34f552..e3f0a7f3131e 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -485,7 +485,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) - if (gpio_request(gpio_nreset, "CS4271 Reset")) + if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) gpio_nreset = -EINVAL; if (gpio_nreset >= 0) { /* Reset codec */ @@ -535,15 +535,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset; - gpio_nreset = cs4271->gpio_nreset; - - if (gpio_is_valid(gpio_nreset)) { + if (gpio_is_valid(cs4271->gpio_nreset)) /* Set codec to the reset state */ - gpio_set_value(gpio_nreset, 0); - gpio_free(gpio_nreset); - } + gpio_set_value(cs4271->gpio_nreset, 0); return 0; }; -- cgit v1.2.3 From d2153a1595ee8235ecf9f9e2d1ac18eee373cbb5 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 13 Nov 2012 10:44:54 +0300 Subject: ALSA: es1968: precedence bug in snd_es1968_tea575x_get_pins() I don't think this works as intended. '|' higher precedence than ?: so the bitwize OR "0 | (val & STR_MOST)" is a no-op. I have re-written it to be more clear. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 50169bcfd903..7266020c16cb 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2581,9 +2581,14 @@ static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 val = inw(io); - - return (val & STR_DATA) ? TEA575X_DATA : 0 | - (val & STR_MOST) ? TEA575X_MOST : 0; + u8 ret; + + ret = 0; + if (val & STR_DATA) + ret |= TEA575X_DATA; + if (val & STR_MOST) + ret |= TEA575X_MOST; + return ret; } static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool output) -- cgit v1.2.3 From effded75e24c7941961d473e4f4babed4c52af3c Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 14 Nov 2012 11:23:54 +0300 Subject: ALSA: fm801: precedence bug in snd_fm801_tea575x_get_pins() There is a precedence bug because | has higher precedence than ?:. This code was cut and pasted and I fixed a similar bug a few days ago. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index cc2e91d15538..c5806f89be1e 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -767,9 +767,14 @@ static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) struct fm801 *chip = tea->private_data; unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); struct snd_fm801_tea575x_gpio gpio = *get_tea575x_gpio(chip); - - return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | - (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; + u8 ret; + + ret = 0; + if (reg & FM801_GPIO_GP(gpio.data)) + ret |= TEA575X_DATA; + if (reg & FM801_GPIO_GP(gpio.most)) + ret |= TEA575X_MOST; + return ret; } static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output) -- cgit v1.2.3 From 10e44239f67d0b6fb74006e61a7e883b8075247a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Nov 2012 11:22:48 +0100 Subject: ALSA: usb-audio: Fix mutex deadlock at disconnection The recent change for USB-audio disconnection race fixes introduced a mutex deadlock again. There is a circular dependency between chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a device is opened during the disconnection operation: A. snd_usb_audio_disconnect() -> card.c::register_mutex -> chip->shutdown_rwsem (write) -> snd_card_disconnect() -> pcm.c::register_mutex -> pcm->open_mutex B. snd_pcm_open() -> pcm->open_mutex -> snd_usb_pcm_open() -> chip->shutdown_rwsem (read) Since the chip->shutdown_rwsem protection in the case A is required only for turning on the chip->shutdown flag and it doesn't have to be taken for the whole operation, we can reduce its window in snd_usb_audio_disconnect(). Reported-by: Jiri Slaby Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 282f0fc9fed1..dbf7999d18b4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -559,9 +559,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, return; card = chip->card; - mutex_lock(®ister_mutex); down_write(&chip->shutdown_rwsem); chip->shutdown = 1; + up_write(&chip->shutdown_rwsem); + + mutex_lock(®ister_mutex); chip->num_interfaces--; if (chip->num_interfaces <= 0) { snd_card_disconnect(card); @@ -582,11 +584,9 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; - up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); snd_card_free_when_closed(card); } else { - up_write(&chip->shutdown_rwsem); mutex_unlock(®ister_mutex); } } -- cgit v1.2.3 From c3c9b370ea4fa2566dfb0c3d88d9f02be0533e7a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Nov 2012 09:26:41 +0900 Subject: ASoC: bells: Fix up git patch application failure It seems git has been getting confused by the very similar contexts for the speaker DAIs and has been applying patches to the wrong places causing all sorts of confusion. Fix this up by hand. Reported-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/samsung/bells.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index b56b9a3c6169..a2ca1567b9e4 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5110-aif3", + .cpu_dai_name = "wm5102-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF -- cgit v1.2.3 From 96e1f18fbb8c146aee9cdad1ebc510b8ccf94b6f Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 15 Nov 2012 11:41:30 +0000 Subject: ASoC: arizona: Fix typo - Swap value in 48k_rates[] and 44k1_rates[] Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c03b65af3059..054967d8bac2 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -268,7 +268,7 @@ EXPORT_SYMBOL_GPL(arizona_out_ev); static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, - 22579200, + 24576000, 49152000, 73728000, 98304000, @@ -278,7 +278,7 @@ static unsigned int arizona_sysclk_48k_rates[] = { static unsigned int arizona_sysclk_44k1_rates[] = { 5644800, 11289600, - 24576000, + 22579200, 45158400, 67737600, 90316800, -- cgit v1.2.3 From e99ddfde6ae0dd2662bb40435696002b590e4057 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 31 Oct 2012 16:35:30 +0100 Subject: ALSA: ua101, usx2y: fix broken MIDI output Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) added autosuspend code to all files making up the snd-usb-audio driver. However, midi.c is part of snd-usb-lib and is also used by other drivers, not all of which support autosuspend. Thus, calls to usb_autopm_get_interface() could fail, and this unexpected error would result in the MIDI output being completely unusable. Make it work by ignoring the error that is expected with drivers that do not support autosuspend. Reported-by: Colin Fletcher Reported-by: Devin Venable Reported-by: Dr Nick Bailey Reported-by: Jannis Achstetter Reported-by: Rui Nuno Capela Cc: Oliver Neukum Cc: 2.6.39+ Signed-off-by: Clemens Ladisch --- sound/usb/midi.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index c83f6143c0eb..eeefbce3873c 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -148,6 +148,7 @@ struct snd_usb_midi_out_endpoint { struct snd_usb_midi_out_endpoint* ep; struct snd_rawmidi_substream *substream; int active; + bool autopm_reference; uint8_t cable; /* cable number << 4 */ uint8_t state; #define STATE_UNKNOWN 0 @@ -1076,7 +1077,8 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) return -ENXIO; } err = usb_autopm_get_interface(umidi->iface); - if (err < 0) + port->autopm_reference = err >= 0; + if (err < 0 && err != -EACCES) return -EIO; substream->runtime->private_data = port; port->state = STATE_UNKNOWN; @@ -1087,9 +1089,11 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct usbmidi_out_port *port = substream->runtime->private_data; substream_open(substream, 0); - usb_autopm_put_interface(umidi->iface); + if (port->autopm_reference) + usb_autopm_put_interface(umidi->iface); return 0; } -- cgit v1.2.3 From 989c3187156ad197ae473fa9d9d506eef9624f12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Nov 2012 14:14:58 +0100 Subject: ALSA: hda - Fix recursive suspend/resume call When the bus reset is performed during the suspend/resume (including the power-saving too), it calls snd_hda_suspend() and snd_hda_resume() again, and deadlocks eventually. For avoiding the recursive call, add a new flag indicating that the PM is being performed, and don't go to the bus reset mode when it's on. Reported-and-tested-by: Julian Wollrath Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 11 +++++++++-- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 70d4848b5cd0..cebe2dfdd984 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -228,7 +228,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, } mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); - if (res && *res == -1 && bus->rirb_error) { + if (!codec->in_pm && res && *res == -1 && bus->rirb_error) { if (bus->response_reset) { snd_printd("hda_codec: resetting BUS due to " "fatal communication error\n"); @@ -238,7 +238,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, goto again; } /* clear reset-flag when the communication gets recovered */ - if (!err) + if (!err || codec->in_pm) bus->response_reset = 0; return err; } @@ -3616,6 +3616,8 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) { unsigned int state; + codec->in_pm = 1; + if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); @@ -3630,6 +3632,7 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) codec->power_transition = 0; codec->power_jiffies = jiffies; spin_unlock(&codec->power_lock); + codec->in_pm = 0; return state; } @@ -3638,6 +3641,8 @@ static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) */ static void hda_call_codec_resume(struct hda_codec *codec) { + codec->in_pm = 1; + /* set as if powered on for avoiding re-entering the resume * in the resume / power-save sequence */ @@ -3656,6 +3661,8 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); } snd_hda_jack_report_sync(codec); + + codec->in_pm = 0; snd_hda_power_down(codec); /* flag down before returning */ } #endif /* CONFIG_PM */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 507fe8a917b6..4f4e545c0f4b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -869,6 +869,7 @@ struct hda_codec { unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ + unsigned int in_pm:1; /* suspend/resume being performed */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- cgit v1.2.3 From 2ea3c6a2c779e5a6487d2b436770232162dfbbe3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Nov 2012 20:03:37 +0100 Subject: ALSA: hda - Limit runtime PM support only to known Intel chips We've got a report that the runtime PM may make the codec the unresponsive on AMD platforms. Since the feature has been tested only on the recent Intel platforms, it's safer to limit the support to such devices for now. This patch adds a new DCAPS bit flag indicating the runtime PM support, and mark it for Intel controllers. Reported-and-tested-by: Julian Wollrath Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 39 ++++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index cd2dbaf1be78..f9d870e554d9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -556,6 +556,12 @@ enum { #define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ +#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ + +/* quirks for Intel PCH */ +#define AZX_DCAPS_INTEL_PCH \ + (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ + AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -2433,6 +2439,9 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up) { struct azx *chip = bus->private_data; + if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + return; + if (power_up) pm_runtime_get_sync(&chip->pci->dev); else @@ -2548,7 +2557,8 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (!power_save_controller) + if (!power_save_controller || + !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return -EAGAIN; azx_stop_chip(chip); @@ -3429,39 +3439,30 @@ static void __devexit azx_remove(struct pci_dev *pci) static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Lynx Point-LP */ { PCI_DEVICE(0x8086, 0x9c21), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0c0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, { PCI_DEVICE(0x8086, 0x0d0c), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY }, + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3 From ae6a5d37725853325a2b3460165fbc5613ce2916 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:17:51 +0000 Subject: ASoC: kirkwood-dma: fix use of virt_to_phys() This is part of a patch found in Rabeeh Khoury's git tree for the cubox. You can not use virt_to_phys() on the address returned from dma_alloc_coherent(); it may not be part of the kernel direct-mapped memory. Fix this to use the DMA address instead. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b9f16598324c..afe193066253 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -178,7 +178,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) } dram = mv_mbus_dram_info(); - addr = virt_to_phys(substream->dma_buffer.area); + addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { prdata->play_stream = substream; kirkwood_dma_conf_mbus_windows(priv->io, -- cgit v1.2.3 From 25ec6bbb63e7eec905d94ccb59cdd54cf22ee618 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:11 +0000 Subject: ASoC: kirkwood-dma: don't ignore other irq causes on error Ignoring the real cause of the interrupt is not a good idea; this behaviour has been observed to bring Dove platforms to silently lockup. Instead, on error fall through to the normal interrupt processing. This is especially important on Dove platforms as errors are handled separately, and allows us to clear down the real cause of the interrupt. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index afe193066253..2ba08148655f 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -71,7 +71,6 @@ static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) printk(KERN_WARNING "%s: got err interrupt 0x%lx\n", __func__, cause); writel(cause, priv->io + KIRKWOOD_ERR_CAUSE); - return IRQ_HANDLED; } /* we've enabled only bytes interrupts ... */ -- cgit v1.2.3 From 2424d458108e275ca736dabc792ee9b6733994c5 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:32 +0000 Subject: ASoC: kirkwood-i2s: fix DCO lock detection This is part of a patch found in Rabeeh Khoury's git tree for the cubox, which is further attributed to Sebastian Hesselbrath. Rather than masking the KIRKWOOD_DCO_SPCR_STATUS register contents against the registers virtual address, let's actually use the bit definition for the locked status, as required in the documentation. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 542538d10ab7..485af80923de 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -95,7 +95,7 @@ static inline void kirkwood_set_dco(void __iomem *io, unsigned long rate) do { cpu_relax(); value = readl(io + KIRKWOOD_DCO_SPCR_STATUS); - value &= KIRKWOOD_DCO_SPCR_STATUS; + value &= KIRKWOOD_DCO_SPCR_STATUS_DCO_LOCK; } while (value == 0); } -- cgit v1.2.3 From 982b604bc56a3da874e489051fc7adb49b1eba65 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:18:52 +0000 Subject: ASoC: kirkwood-i2s: fix DMA underruns Stress testing the driver with multiple start/stop events causes kirkwood-dma to report underrun errors (which used to cause the kernel to lock up solidly). This is because kirkwood-i2s is not respecting the restrictions imposed on clearing the 'pause' bit. Follow what the spec says; the busy bit must be read as being clear twice before the pause bit can be released. This solves the underruns. However, it has been noticed that the busy bit occasionally does not clear itself, hence the waiting is bounded to 5ms maximum to avoid a new reason for the kernel to lockup. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 67 ++++++++++++++++++++++----------------- 1 file changed, 38 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 485af80923de..826306dfb72b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -180,67 +180,76 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); - unsigned long value; - - /* - * specs says KIRKWOOD_PLAYCTL must be read 2 times before - * changing it. So read 1 time here and 1 later. - */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); + uint32_t ctl, value; + + ctl = readl(priv->io + KIRKWOOD_PLAYCTL); + if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + unsigned timeout = 5000; + /* + * The Armada510 spec says that if we enter pause mode, the + * busy bit must be read back as clear _twice_. Make sure + * we respect that otherwise we get DMA underruns. + */ + do { + value = ctl; + ctl = readl(priv->io + KIRKWOOD_PLAYCTL); + if (!((ctl | value) & KIRKWOOD_PLAYCTL_PLAY_BUSY)) + break; + udelay(1); + } while (timeout--); + + if ((ctl | value) & KIRKWOOD_PLAYCTL_PLAY_BUSY) + dev_notice(dai->dev, "timed out waiting for busy to deassert: %08x\n", + ctl); + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* stop audio, enable interrupts */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); /* configure audio & enable i2s playback */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~KIRKWOOD_PLAYCTL_BURST_MASK; - value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE + ctl &= ~KIRKWOOD_PLAYCTL_BURST_MASK; + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_EN); if (priv->burst == 32) - value |= KIRKWOOD_PLAYCTL_BURST_32; + ctl |= KIRKWOOD_PLAYCTL_BURST_32; else - value |= KIRKWOOD_PLAYCTL_BURST_128; - value |= KIRKWOOD_PLAYCTL_I2S_EN; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_BURST_128; + ctl |= KIRKWOOD_PLAYCTL_I2S_EN; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); value &= ~KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); /* disable all playbacks */ - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN); + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - value = readl(priv->io + KIRKWOOD_PLAYCTL); - value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); - writel(value, priv->io + KIRKWOOD_PLAYCTL); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; default: -- cgit v1.2.3 From 3ccdf5bbdf5f2488e4a36692d055ba9c43ae6717 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 20 Nov 2012 12:19:13 +0000 Subject: ASoC: kirkwood-i2s: more pause-mode fixes Don't even momentarily set the pause status when starting the channel; if we do, we should check the busy bit to ensure that we comply with the spec. In any case, it isn't necessary; we will not active on a START event so there is no need to pause the DMA. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 826306dfb72b..1d5db484d2df 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -205,10 +205,6 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - /* stop audio, enable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE; - writel(ctl, priv->io + KIRKWOOD_PLAYCTL); - value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); @@ -269,11 +265,6 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - /* stop audio, enable interrupts */ - value = readl(priv->io + KIRKWOOD_RECCTL); - value |= KIRKWOOD_RECCTL_PAUSE; - writel(value, priv->io + KIRKWOOD_RECCTL); - value = readl(priv->io + KIRKWOOD_INT_MASK); value |= KIRKWOOD_INT_CAUSE_REC_BYTES; writel(value, priv->io + KIRKWOOD_INT_MASK); -- cgit v1.2.3 From af02dde8a609d8d071c4b31a82df811a55690a4a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 21 Nov 2012 08:57:58 +0100 Subject: ALSA: hda - Add support for Realtek ALC292 We found a new codec ID 292, and that just a simple quirk would enable sound output/input on this ALC292 chip. BugLink: https://bugs.launchpad.net/bugs/1081466 Cc: stable@vger.kernel.org Tested-by: Acelan Kao Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 68fd49294b26..ad68d223f8af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7065,6 +7065,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, + { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit v1.2.3 From 34c3d1926bdaf45d3a891dd577482abcdd9faa34 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 21 Nov 2012 10:03:10 +0100 Subject: ALSA: hda - Cirrus: Correctly clear line_out_pins when moving to speaker If this array is not cleared, the jack related code later might fail to create "Internal Speaker Phantom Jack" on Dell Inspiron 3420 and Dell Vostro 2420. BugLink: https://bugs.launchpad.net/bugs/1076840 Cc: stable@vger.kernel.org (3.6+) Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d5f3a26d608d..3bcb67172358 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -466,6 +466,7 @@ static int parse_output(struct hda_codec *codec) memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins)); cfg->line_outs = 0; + memset(cfg->line_out_pins, 0, sizeof(cfg->line_out_pins)); } return 0; -- cgit v1.2.3 From 947d299686aa9cc8aecf749d54e8475c6e498956 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 22 Nov 2012 20:27:59 +0100 Subject: ALSA: snd-usb: properly initialize the sync endpoint Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio driver which causes the code to not initialize the sync endpoint from configure_endpoint(). Reported-by: Jeffrey Barish Signed-off-by: Daniel Mack Cc: stable@kernel.org [3.5+] Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5c12a3fe8c3e..ef6fa24fc473 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -459,7 +459,7 @@ static int configure_endpoint(struct snd_usb_substream *subs) return ret; if (subs->sync_endpoint) - ret = snd_usb_endpoint_set_params(subs->data_endpoint, + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, subs->pcm_format, subs->channels, subs->period_bytes, -- cgit v1.2.3 From d846b17475d52f037437d125cd19c28f1d36e4f0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 24 Nov 2012 11:58:24 +0100 Subject: ALSA: hda - Fix build without CONFIG_PM I forgot this again... codec->in_pm is in #ifdef CONFIG_PM Reported-by: Markus Trippelsdorf Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cebe2dfdd984..d010de12335e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -95,6 +95,7 @@ int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); #ifdef CONFIG_PM +#define codec_in_pm(codec) ((codec)->in_pm) static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) @@ -104,6 +105,7 @@ static inline void hda_call_pm_notify(struct hda_bus *bus, bool power_up) bus->ops.pm_notify(bus, power_up); } #else +#define codec_in_pm(codec) 0 static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 #define hda_call_pm_notify(bus, state) {} @@ -228,7 +230,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, } mutex_unlock(&bus->cmd_mutex); snd_hda_power_down(codec); - if (!codec->in_pm && res && *res == -1 && bus->rirb_error) { + if (!codec_in_pm(codec) && res && *res == -1 && bus->rirb_error) { if (bus->response_reset) { snd_printd("hda_codec: resetting BUS due to " "fatal communication error\n"); @@ -238,7 +240,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, goto again; } /* clear reset-flag when the communication gets recovered */ - if (!err || codec->in_pm) + if (!err || codec_in_pm(codec)) bus->response_reset = 0; return err; } -- cgit v1.2.3