From 2a67e796da38dff41e5b2aa9352c81189407acb9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:06 +0000 Subject: ASoC: wm5102: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ce9c8e14d4bd..5e03a83aecaa 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +698,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +711,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +724,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit From 552694c65b1c563dbdbda4082527774a373ae720 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:07 +0000 Subject: ASoC: wm5110: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2c3c962d9a85..7c2b0d669d29 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,15 +247,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -267,6 +260,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -278,6 +273,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -289,6 +286,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit From fb0def0ceda7b2a79be978093704c1c71e26ba22 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:08 +0000 Subject: ASoC: wm8997: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 555115ee2159..4a382495976c 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -170,15 +170,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -190,6 +183,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -201,6 +196,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -212,6 +209,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit From ddbc5efed0f9064287acead56bbf0dce3ca28ee2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 22 Jan 2014 10:09:11 +0000 Subject: ASoC: wm_adsp: Add debug print to note that the DSP has shutdown It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 444626fcab40..f9fd56444a14 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1679,6 +1679,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_dbg(dsp, "Shutdown complete\n"); break; default: -- cgit From 1371105731a7c72168d0e464a51203fec829390b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:34 +0000 Subject: ASoC: wm5102: Correct typo in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5e03a83aecaa..ebffe81daa1d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,7 +685,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -698,7 +698,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -711,7 +711,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -724,7 +724,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit From 3c379ef97e7f6cd305a4873150319c2355b4316d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:35 +0000 Subject: ASoC: wm5110: Correct type in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7c2b0d669d29..4de2bf16dc74 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,7 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -260,7 +260,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -273,7 +273,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -286,7 +286,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit From 2b14cd3af9d98ce135e503e91e3973bdd5e4baeb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:36 +0000 Subject: ASoC: wm8997: Correct typo in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4a382495976c..6107108228b6 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -170,7 +170,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -183,7 +183,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -196,7 +196,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -209,7 +209,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit From b224e9b857438afbd802f47008ab36863f71d8d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Feb 2014 14:48:22 +0100 Subject: ASoC: cs4271: Remove outdated comment Commit 1b1861ead ("ASoC: cs4271: convert to direct regmap API usage") removed the bus_type field from the cs4271_private struct, but left the comment that described the field in there. This patch removes the comment. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd93dc74..f7bbe6fdba67 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -159,7 +159,6 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) } struct cs4271_private { - /* SND_SOC_I2C or SND_SOC_SPI */ unsigned int mclk; bool master; bool deemph; -- cgit From 6e84b9768dfb299a9881895b331e3e532041fae4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Feb 2014 20:55:31 +0100 Subject: ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates SNDRV_PCM_RATE_KNOT means that the device can support rates that can not be expressed using the rate bits. The driver will provide a list of those rates specified through constraints. Any rate bits that are set in the rates mask will be ignored. So setting other rate bits besides SNDRV_PCM_RATE_KNOT wont have any effect, but might be confusing to the casual reader, so remove them. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..7cae046c7dd0 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) - #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "XSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "XSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "ASP Playback", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "ASP Capture", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "VSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "VSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, -- cgit From 096ae5444b8600bbee0501b01987094657a1458e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:32 +0100 Subject: ASoC: cs42l73: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. While we are at it also change the type for the rates array from u32 to unsigned int. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7cae046c7dd0..69c8e2de7d0e 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1108,7 +1108,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -static u32 cs42l73_asrc_rates[] = { +static const unsigned int cs42l73_asrc_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; @@ -1241,7 +1241,7 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) 0x7F, tristate << 7); } -static struct snd_pcm_hw_constraint_list constraints_12_24 = { +static const struct snd_pcm_hw_constraint_list constraints_12_24 = { .count = ARRAY_SIZE(cs42l73_asrc_rates), .list = cs42l73_asrc_rates, }; -- cgit From dfd72a68aa0f6cf87575f3181319bde8a2d4c01b Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Thu, 30 Jan 2014 18:14:05 +0100 Subject: ASoC: cs42l51: add Device Tree binding to cs42l51 This commit adds a trivial Device Tree binding to the I2C-based cs42l51 sound codec, so that it can be used from Device Tree based platforms. Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..824cdf4d4974 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -600,10 +600,17 @@ static const struct i2c_device_id cs42l51_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_id); +static const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51-codec", .owner = THIS_MODULE, + .of_match_table = cs42l51_of_match, }, .id_table = cs42l51_id, .probe = cs42l51_i2c_probe, -- cgit From 5be736442ed94217c6521ae0c948abab995f281f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:21:16 +0000 Subject: ASoC: cs42l51: Don't log if we fail to allocate memory The VM subsystem already logs quite loudly if we run out of memory so don't bother here. Signed-off-by: Mark Brown Acked-by: Brian Austin --- sound/soc/codecs/cs42l51.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 824cdf4d4974..b11079b38f15 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -574,10 +574,8 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); - if (!cs42l51) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l51) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs42l51); cs42l51->control_type = SND_SOC_I2C; -- cgit From da071489762499a3635cb3563d32792cea20c087 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:24:31 +0000 Subject: ASoC: cs42l51: Convert to direct regmap API usage As part of phasing out the ASoC level register I/O code (which is now just a thin wrapper around regmap anyway) convert the cs42l51 driver to use the regmap API directly. We now no longer initialise the cache from hardware at startup, the regmap caches are smart enough to understand which registers are actually cached and read on demand. This should have no visible effect on the system. Signed-off-by: Mark Brown Acked-by: Brian Austin --- sound/soc/codecs/cs42l51.c | 59 ++++++++++++++++++++-------------------------- 1 file changed, 25 insertions(+), 34 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b11079b38f15..e53c8714591f 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "cs42l51.h" @@ -40,7 +41,6 @@ enum master_slave_mode { }; struct cs42l51_private { - enum snd_soc_control_type control_type; unsigned int mclk; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; @@ -52,24 +52,6 @@ struct cs42l51_private { SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -static int cs42l51_fill_cache(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache + 1; - struct i2c_client *i2c_client = to_i2c_client(codec->dev); - s32 length; - - length = i2c_smbus_read_i2c_block_data(i2c_client, - CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache); - if (length != CS42L51_NUMREGS) { - dev_err(&i2c_client->dev, - "I2C read failure, addr=0x%x (ret=%d vs %d)\n", - i2c_client->addr, length, CS42L51_NUMREGS); - return -EIO; - } - - return 0; -} - static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -508,13 +490,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; - ret = cs42l51_fill_cache(codec); - if (ret < 0) { - dev_err(codec->dev, "failed to fill register cache\n"); - return ret; - } - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l51->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -538,8 +514,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS + 1, - .reg_word_size = sizeof(u8), .controls = cs42l51_snd_controls, .num_controls = ARRAY_SIZE(cs42l51_snd_controls), @@ -549,28 +523,46 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; +static const struct regmap_config cs42l51_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L51_CHARGE_FREQ, + .cache_type = REGCACHE_RBTREE, +}; + static int cs42l51_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l51_private *cs42l51; + struct regmap *regmap; + unsigned int val; int ret; + regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c_client->dev, "Failed to create regmap: %d\n", + ret); + return ret; + } + /* Verify that we have a CS42L51 */ - ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID); + ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { dev_err(&i2c_client->dev, "failed to read I2C\n"); goto error; } - if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && - (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { - dev_err(&i2c_client->dev, "Invalid chip id\n"); + if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && + (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { + dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val); ret = -ENODEV; goto error; } dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", - ret & 7); + val & 7); cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); @@ -578,7 +570,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; i2c_set_clientdata(i2c_client, cs42l51); - cs42l51->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); -- cgit From 048d4ff81f1cf26b3f3627a9a69d35aff7898bb3 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Mon, 10 Feb 2014 14:09:45 +0800 Subject: ASoC: atmel_ssc_dai: make option to choose clock When SSC works in slave mode, according to the hardware design, the clock can get from TK pin, also can get from RK pin. So, add one parameter to choose where the clock from. Signed-off-by: Bo Shen Acked-by: Nicolas Ferre Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1ead3c977a51..de433cfd044c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -341,6 +341,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, { int id = dai->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct ssc_device *ssc = ssc_p->ssc; struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; @@ -466,7 +467,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_START, start_event) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) - | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_PIN : SSC_CKS_CLOCK); rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) @@ -481,7 +483,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_START, start_event) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) - | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_CLOCK : SSC_CKS_PIN); tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) @@ -550,7 +553,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_START, SSC_START_RISING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) - | SSC_BF(RCMR_CKS, SSC_CKS_PIN); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_PIN : SSC_CKS_CLOCK); rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) @@ -565,7 +569,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_START, SSC_START_RISING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) - | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_CLOCK : SSC_CKS_PIN); tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) -- cgit From 423f0c4a3d32cc83dff204324f59aecb4516f3cf Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 18 Feb 2014 14:32:48 +0530 Subject: ASoC: cs42l51: Remove unused variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ‘cs42l51’ is not used. Remove it. Signed-off-by: Sachin Kamat Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e53c8714591f..3eab46020a30 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -487,7 +487,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { - struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); -- cgit From d6cf89ee07cbfd980f189cc12ae924c811b00ee4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 19 Feb 2014 14:05:54 +0100 Subject: ASoC: cs4271: claim reset GPIO in bus probe function Move the GPIO acquisition from the codec to the bus probe functions. Signed-off-by: Daniel Mack Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 60 +++++++++++++++++++++++++++++++---------------- 1 file changed, 40 insertions(+), 20 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f7bbe6fdba67..96c309777208 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -539,14 +539,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; - int gpio_nreset = -EINVAL; bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { - gpio_nreset = of_get_named_gpio(codec->dev->of_node, - "reset-gpio", 0); - if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; @@ -558,27 +554,19 @@ static int cs4271_probe(struct snd_soc_codec *codec) #endif if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_nreset >= 0) - if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) - gpio_nreset = -EINVAL; - if (gpio_nreset >= 0) { + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ - gpio_direction_output(gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); udelay(1); - gpio_set_value(gpio_nreset, 1); + gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ udelay(1); } - cs4271->gpio_nreset = gpio_nreset; - ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); @@ -640,14 +628,45 @@ static const struct regmap_config cs4271_spi_regmap = { .volatile_reg = cs4271_volatile_reg, }; -static int cs4271_spi_probe(struct spi_device *spi) +static int cs4271_common_probe(struct device *dev, + struct cs4271_private **c) { + struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; - cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) return -ENOMEM; + if (of_match_device(cs4271_dt_ids, dev)) + cs4271->gpio_nreset = + of_get_named_gpio(dev->of_node, "reset-gpio", 0); + + if (cs4271plat) + cs4271->gpio_nreset = cs4271plat->gpio_nreset; + + if (gpio_is_valid(cs4271->gpio_nreset)) { + int ret; + + ret = devm_gpio_request(dev, cs4271->gpio_nreset, + "CS4271 Reset"); + if (ret < 0) + return ret; + } + + *c = cs4271; + return 0; +} + +static int cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + int ret; + + ret = cs4271_common_probe(&spi->dev, &cs4271); + if (ret < 0) + return ret; + spi_set_drvdata(spi, cs4271); cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); if (IS_ERR(cs4271->regmap)) @@ -697,10 +716,11 @@ static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&client->dev, &cs4271); + if (ret < 0) + return ret; i2c_set_clientdata(client, cs4271); cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); -- cgit From 4682a0a2b8e5d9633727276455f445e0b402767c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:05:01 +0100 Subject: ASoC: cs42l52: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 92 ++++++++++++++++++++++------------------------ 1 file changed, 43 insertions(+), 49 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..be455ea5f2fe 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -210,13 +210,11 @@ static const char * const cs42l52_adca_text[] = { static const char * const cs42l52_adcb_text[] = { "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; -static const struct soc_enum adca_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, - ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); +static SOC_ENUM_SINGLE_DECL(adca_enum, + CS42L52_ADC_PGA_A, 5, cs42l52_adca_text); -static const struct soc_enum adcb_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, - ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); +static SOC_ENUM_SINGLE_DECL(adcb_enum, + CS42L52_ADC_PGA_B, 5, cs42l52_adcb_text); static const struct snd_kcontrol_new adca_mux = SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); @@ -229,26 +227,22 @@ static const char * const mic_bias_level_text[] = { "0.8 +VA", "0.83 +VA", "0.91 +VA" }; -static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, - ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); +static SOC_ENUM_SINGLE_DECL(mic_bias_level_enum, + CS42L52_IFACE_CTL2, 0, mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" }; -static const struct soc_enum mica_enum = - SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(mica_enum, + CS42L52_MICA_CTL, 5, cs42l52_mic_text); -static const struct soc_enum micb_enum = - SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(micb_enum, + CS42L52_MICB_CTL, 5, cs42l52_mic_text); static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; -static const struct soc_enum digital_output_mux_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, - ARRAY_SIZE(digital_output_mux_text), - digital_output_mux_text); +static SOC_ENUM_SINGLE_DECL(digital_output_mux_enum, + CS42L52_ADC_MISC_CTL, 6, + digital_output_mux_text); static const struct snd_kcontrol_new digital_output_mux = SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); @@ -258,18 +252,18 @@ static const char * const hp_gain_num_text[] = { "0.7099", "0.8399", "1.000", "1.1430" }; -static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, - ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); +static SOC_ENUM_SINGLE_DECL(hp_gain_enum, + CS42L52_PB_CTL1, 5, + hp_gain_num_text); static const char * const beep_pitch_text[] = { "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" }; -static const struct soc_enum beep_pitch_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, - ARRAY_SIZE(beep_pitch_text), beep_pitch_text); +static SOC_ENUM_SINGLE_DECL(beep_pitch_enum, + CS42L52_BEEP_FREQ, 4, + beep_pitch_text); static const char * const beep_ontime_text[] = { "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", @@ -277,66 +271,66 @@ static const char * const beep_ontime_text[] = { "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" }; -static const struct soc_enum beep_ontime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, - ARRAY_SIZE(beep_ontime_text), beep_ontime_text); +static SOC_ENUM_SINGLE_DECL(beep_ontime_enum, + CS42L52_BEEP_FREQ, 0, + beep_ontime_text); static const char * const beep_offtime_text[] = { "1.23 s", "2.58 s", "3.90 s", "5.20 s", "6.60 s", "8.05 s", "9.35 s", "10.80 s" }; -static const struct soc_enum beep_offtime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, - ARRAY_SIZE(beep_offtime_text), beep_offtime_text); +static SOC_ENUM_SINGLE_DECL(beep_offtime_enum, + CS42L52_BEEP_VOL, 5, + beep_offtime_text); static const char * const beep_config_text[] = { "Off", "Single", "Multiple", "Continuous" }; -static const struct soc_enum beep_config_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, - ARRAY_SIZE(beep_config_text), beep_config_text); +static SOC_ENUM_SINGLE_DECL(beep_config_enum, + CS42L52_BEEP_TONE_CTL, 6, + beep_config_text); static const char * const beep_bass_text[] = { "50 Hz", "100 Hz", "200 Hz", "250 Hz" }; -static const struct soc_enum beep_bass_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, - ARRAY_SIZE(beep_bass_text), beep_bass_text); +static SOC_ENUM_SINGLE_DECL(beep_bass_enum, + CS42L52_BEEP_TONE_CTL, 1, + beep_bass_text); static const char * const beep_treble_text[] = { "5 kHz", "7 kHz", "10 kHz", " 15 kHz" }; -static const struct soc_enum beep_treble_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, - ARRAY_SIZE(beep_treble_text), beep_treble_text); +static SOC_ENUM_SINGLE_DECL(beep_treble_enum, + CS42L52_BEEP_TONE_CTL, 3, + beep_treble_text); static const char * const ng_threshold_text[] = { "-34dB", "-37dB", "-40dB", "-43dB", "-46dB", "-52dB", "-58dB", "-64dB" }; -static const struct soc_enum ng_threshold_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, - ARRAY_SIZE(ng_threshold_text), ng_threshold_text); +static SOC_ENUM_SINGLE_DECL(ng_threshold_enum, + CS42L52_NOISE_GATE_CTL, 2, + ng_threshold_text); static const char * const cs42l52_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms"}; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, - ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L52_NOISE_GATE_CTL, 0, + cs42l52_ng_delay_text); static const char * const cs42l52_ng_type_text[] = { "Apply Specific", "Apply All" }; -static const struct soc_enum ng_type_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, - ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); +static SOC_ENUM_SINGLE_DECL(ng_type_enum, + CS42L52_NOISE_GATE_CTL, 6, + cs42l52_ng_type_text); static const char * const left_swap_text[] = { "Left", "LR 2", "Right"}; -- cgit From e34042d850a7117b9acafeabd50287d5d8e61849 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:06:00 +0100 Subject: ASoC: da7210: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e62e294a8033..01e55fc72307 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -307,29 +307,29 @@ static const char * const da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; -static const struct soc_enum da7210_dac_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_hpf_cutoff, + DA7210_DAC_HPF, 0, da7210_hpf_cutoff_txt); -static const struct soc_enum da7210_adc_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_hpf_cutoff, + DA7210_ADC_HPF, 0, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ static const char * const da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da7210_dac_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_vf_cutoff, + DA7210_DAC_HPF, 4, da7210_vf_cutoff_txt); -static const struct soc_enum da7210_adc_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_vf_cutoff, + DA7210_ADC_HPF, 4, da7210_vf_cutoff_txt); static const char *da7210_hp_mode_txt[] = { "Class H", "Class G" }; -static const struct soc_enum da7210_hp_mode_sel = - SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +static SOC_ENUM_SINGLE_DECL(da7210_hp_mode_sel, + DA7210_HP_CFG, 0, da7210_hp_mode_txt); /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, -- cgit From 52a5b545bc09ebc7b1e4a55d765ccb76286ca48d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:57:55 +0100 Subject: ASoC: cs42l73: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 69c8e2de7d0e..06f429184821 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -278,13 +278,13 @@ static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; -static const struct soc_enum pgaa_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, - ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); +static SOC_ENUM_SINGLE_DECL(pgaa_enum, + CS42L73_ADCIPC, 3, + cs42l73_pgaa_text); -static const struct soc_enum pgab_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, - ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); +static SOC_ENUM_SINGLE_DECL(pgab_enum, + CS42L73_ADCIPC, 7, + cs42l73_pgab_text); static const struct snd_kcontrol_new pgaa_mux = SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); @@ -309,9 +309,9 @@ static const struct snd_kcontrol_new input_right_mixer[] = { static const char * const cs42l73_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms" }; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, - ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L73_NGCAB, 0, + cs42l73_ng_delay_text); static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -357,19 +357,19 @@ static const struct snd_kcontrol_new esl_xsp_mixer = static const char * const cs42l73_ip_swap_text[] = { "Stereo", "Mono A", "Mono B", "Swap A-B"}; -static const struct soc_enum ip_swap_enum = - SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, - ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); +static SOC_ENUM_SINGLE_DECL(ip_swap_enum, + CS42L73_MIOPC, 6, + cs42l73_ip_swap_text); static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; -static const struct soc_enum vsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(vsp_output_mux_enum, + CS42L73_MIXERCTL, 5, + cs42l73_spo_mixer_text); -static const struct soc_enum xsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum, + CS42L73_MIXERCTL, 4, + cs42l73_spo_mixer_text); static const struct snd_kcontrol_new vsp_output_mux = SOC_DAPM_ENUM("Route", vsp_output_mux_enum); -- cgit From daf152a21d361967bd9d2d7a976c125de842d589 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:49:02 +0100 Subject: ASoC: wm5102: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ... to make clear the meaning of the argument. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ebffe81daa1d..293dffcb1d2f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -622,13 +622,16 @@ static const unsigned int wm5102_osr_val[] = { static const struct soc_enum wm5102_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT2_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), }; -- cgit From 347e5512642da44d15bf8b48ee0fe196b37a78f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:49:56 +0100 Subject: ASoC: wm8997: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ... to make clear the meaning of the argument. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 6107108228b6..4e6442ce9a2a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -123,10 +123,12 @@ static const unsigned int wm8997_osr_val[] = { static const struct soc_enum wm8997_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), }; -- cgit From 17cb37aafdc11b875b915292ae21ac3a4f1425a7 Mon Sep 17 00:00:00 2001 From: Alexander Shiyan Date: Tue, 25 Feb 2014 22:01:54 +0400 Subject: ASoC: cirrus: Remove excess dependencies on SND_SOC Configuration for Cirrus Logic audio support is included only if SND_SOC symbol selected, so no reason to check it once more. Signed-off-by: Alexander Shiyan Signed-off-by: Mark Brown --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 06f938deda15..c872dacbab98 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC + depends on ARCH_EP93XX || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- cgit From 055bbe2df957343fece60fe1f60553a9c1005217 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 1 Mar 2014 13:45:32 +0100 Subject: ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm5192, wm5110 and wm8997 drivers for this by using arizona->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffcb1d2f..34109050ceed 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -582,7 +582,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf16dc74..d7bf8848174a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -136,7 +136,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442ce9a2a..e10f44d7fdb7 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -86,7 +86,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; -- cgit From 9c369c6e885599818d98ff7130d6ef62ce6ae8d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 5 Mar 2014 17:53:31 +0100 Subject: ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cs4271_common_probe() is called from cs4271_i2c_probe() but defined in CONFIG_SPI_MASTER block, thus it results in a build error when CONFIG_SPI_MASTER=n: sound/soc/codecs/cs4271.c:721:2: error: implicit declaration of function ‘cs4271_common_probe’ [-Werror=implicit-function-declaration] Move the function out of #if block. Fixes: d6cf89ee07cb ('ASoC: cs4271: claim reset GPIO in bus probe function') Signed-off-by: Takashi Iwai Acked-by: Daniel Mack Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 96c309777208..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -612,22 +612,6 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, - .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { @@ -658,6 +642,22 @@ static int cs4271_common_probe(struct device *dev, return 0; } +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; -- cgit From c1a7898d655fd265feefcf6fe82ab0096e6d078e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Mar 2014 14:28:16 +0000 Subject: ASoC: wm_adsp: Split firmware load into smaller chunks The firmware files can be quite large and allocating the whole firmware a single DMA safe buffer can be problematic if the system is under a high memory load. Ease the requirements slightly by writing the firmware out in page sized chunks. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 48 ++++++++++++++++++++++++++++++---------------- 1 file changed, 31 insertions(+), 17 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9fd56444a14..937af6f31ffa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = wm_adsp_buf_alloc(region->data, - le32_to_cpu(region->len), - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } + size_t to_write = PAGE_SIZE; + size_t remain = le32_to_cpu(region->len); + const u8 *data = region->data; + + while (remain > 0) { + if (remain < PAGE_SIZE) + to_write = remain; + + buf = wm_adsp_buf_alloc(data, + to_write, + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", - file, regions, - le32_to_cpu(region->len), offset, - region_name, ret); - goto out_fw; + ret = regmap_raw_write_async(regmap, reg, + buf->buf, + to_write); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + file, regions, + to_write, offset, + region_name, ret); + goto out_fw; + } + + data += to_write; + reg += to_write / 2; + remain -= to_write; } } -- cgit From fab800cc33e98378336faf75688ea0961eac21b6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 6 Mar 2014 10:00:18 +0000 Subject: ASoC: wm_adsp: Correct type specifier in printf Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 937af6f31ffa..bb5f7b4e3ebb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -706,7 +706,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) to_write); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", file, regions, to_write, offset, region_name, ret); -- cgit From da28ed585b26dc6eb0c8d897a9b842a86dd6a659 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:17 +0000 Subject: ASoC: arizona: An OUTDIV of 1 is not valid, avoid this One is not a valid value for the OUTDIV start searching at 2 instead. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fee8f13..d90804686e4e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1406,7 +1406,7 @@ static int arizona_calc_fll(struct arizona_fll *fll, Fref /= div; /* Fvco should be over the targt; don't check the upper bound */ - div = 1; + div = 2; while (Fout * div < 90000000 * fll->vco_mult) { div++; if (div > 7) { -- cgit From 87383ac5a73ff34c60d3ea483bf24cabb27fb522 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:18 +0000 Subject: ASoC: arizona: Add defines for FLL configuration constants Improve readability by adding defines for some of the constants associated with FLL configuration. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d90804686e4e..3d4408db075f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -53,6 +53,12 @@ #define ARIZONA_AIF_RX_ENABLES 0x1A #define ARIZONA_AIF_FORCE_WRITE 0x1B +#define ARIZONA_FLL_MAX_FREF 13500000 +#define ARIZONA_FLL_MIN_FVCO 90000000 +#define ARIZONA_FLL_MAX_REFDIV 8 +#define ARIZONA_FLL_MIN_OUTDIV 2 +#define ARIZONA_FLL_MAX_OUTDIV 7 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -1390,11 +1396,11 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Fref must be <=13.5MHz */ div = 1; cfg->refdiv = 0; - while ((Fref / div) > 13500000) { + while ((Fref / div) > ARIZONA_FLL_MAX_FREF) { div *= 2; cfg->refdiv++; - if (div > 8) { + if (div > ARIZONA_FLL_MAX_REFDIV) { arizona_fll_err(fll, "Can't scale %dMHz in to <=13.5MHz\n", Fref); @@ -1406,10 +1412,10 @@ static int arizona_calc_fll(struct arizona_fll *fll, Fref /= div; /* Fvco should be over the targt; don't check the upper bound */ - div = 2; - while (Fout * div < 90000000 * fll->vco_mult) { + div = ARIZONA_FLL_MIN_OUTDIV; + while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; - if (div > 7) { + if (div > ARIZONA_FLL_MAX_OUTDIV) { arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", Fout); return -EINVAL; -- cgit From 61719db8141acde1a6293bbbddc733655defcc3c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:19 +0000 Subject: ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll Since we know in arizona_apply_fll if we are setting the sync or ref path there is no need to set the outdiv seperately anymore. This patch moves this from arizona_enable_fll to arizona_apply_fll. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 28 ++++++++++++---------------- 1 file changed, 12 insertions(+), 16 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 3d4408db075f..9afd8c41d143 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1502,14 +1502,18 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); - if (sync) - regmap_update_bits_async(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - else - regmap_update_bits_async(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + if (sync) { + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } else { + regmap_update_bits(arizona->regmap, base + 0x5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } regmap_update_bits_async(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, @@ -1546,10 +1550,6 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); if (fll->sync_src >= 0) { @@ -1558,10 +1558,6 @@ static void arizona_enable_fll(struct arizona_fll *fll, use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); -- cgit From 23f785a8bc33a98c4c384a653b9bff9c0cc3591d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:20 +0000 Subject: ASoC: arizona: Move calculation of FLL configuration Currently the FLL configuration is calculated before it is known which FLL path the configuration will be applied to. Newer versions of the IP have differences in the configuration required for each FLL path, which makes it complicated to calculate the FLL configuration in advance. This patch simply checks the validity of a requested input and output frequency before we know which FLL path they will be applied to and saves the actual calculation of the configuration until we know where the settings will be applied. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 79 ++++++++++++++++++++++++++-------------------- 1 file changed, 44 insertions(+), 35 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9afd8c41d143..7398c69192cb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1383,6 +1383,29 @@ struct arizona_fll_cfg { int gain; }; +static int arizona_validate_fll(struct arizona_fll *fll, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int Fvco_min; + + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + + Fvco_min = ARIZONA_FLL_MIN_FVCO * fll->vco_mult; + if (Fout * ARIZONA_FLL_MAX_OUTDIV < Fvco_min) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + + return 0; +} + static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, unsigned int Fref, @@ -1400,12 +1423,8 @@ static int arizona_calc_fll(struct arizona_fll *fll, div *= 2; cfg->refdiv++; - if (div > ARIZONA_FLL_MAX_REFDIV) { - arizona_fll_err(fll, - "Can't scale %dMHz in to <=13.5MHz\n", - Fref); + if (div > ARIZONA_FLL_MAX_REFDIV) return -EINVAL; - } } /* Apply the division for our remaining calculations */ @@ -1415,11 +1434,8 @@ static int arizona_calc_fll(struct arizona_fll *fll, div = ARIZONA_FLL_MIN_OUTDIV; while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; - if (div > ARIZONA_FLL_MAX_OUTDIV) { - arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", - Fout); + if (div > ARIZONA_FLL_MAX_OUTDIV) return -EINVAL; - } } target = Fout * div / fll->vco_mult; cfg->outdiv = div; @@ -1536,13 +1552,12 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll, - struct arizona_fll_cfg *ref, - struct arizona_fll_cfg *sync) +static void arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + struct arizona_fll_cfg cfg; /* * If we have both REFCLK and SYNCCLK then enable both, @@ -1550,15 +1565,21 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + arizona_calc_fll(fll, &cfg, fll->ref_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_apply_fll(arizona, fll->base + 0x10, sync, + arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_apply_fll(arizona, fll->base, sync, + arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); regmap_update_bits_async(arizona->regmap, fll->base + 0x11, @@ -1620,32 +1641,22 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout) { - if (Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; - } - - if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, - fll->fout); - if (ret != 0) - return ret; - } + if (fll->fout && Fref > 0) { + ret = arizona_validate_fll(fll, Fref, fll->fout); + if (ret != 0) + return ret; } fll->ref_src = source; fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } return 0; @@ -1655,7 +1666,6 @@ EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->sync_src == source && @@ -1664,13 +1674,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (Fout) { if (fll->ref_src >= 0) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, - Fout); + ret = arizona_validate_fll(fll, fll->ref_freq, Fout); if (ret != 0) return ret; } - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + ret = arizona_validate_fll(fll, Fref, Fout); if (ret != 0) return ret; } @@ -1680,7 +1689,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->fout = Fout; if (Fout) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } else { arizona_disable_fll(fll); } -- cgit From 8ccefcd265b486186c94ea70c77511e7c570347d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:21 +0000 Subject: ASoC: arizona: Don't pass Fout into arizona_calc_fll As we now calculate the FLL configuration at a later stage in the process the fout member of the FLL structure will contain the desired Fout frequency so no need to pass this in seperately. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7398c69192cb..7b1354ae337b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1408,13 +1408,12 @@ static int arizona_validate_fll(struct arizona_fll *fll, static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, - unsigned int Fref, - unsigned int Fout) + unsigned int Fref) { unsigned int target, div, gcd_fll; int i, ratio; - arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout); /* Fref must be <=13.5MHz */ div = 1; @@ -1432,12 +1431,12 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Fvco should be over the targt; don't check the upper bound */ div = ARIZONA_FLL_MIN_OUTDIV; - while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { + while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; if (div > ARIZONA_FLL_MAX_OUTDIV) return -EINVAL; } - target = Fout * div / fll->vco_mult; + target = fll->fout * div / fll->vco_mult; cfg->outdiv = div; arizona_fll_dbg(fll, "Fvco=%dHz\n", target); @@ -1565,19 +1564,19 @@ static void arizona_enable_fll(struct arizona_fll *fll) */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_calc_fll(fll, &cfg, fll->ref_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->ref_freq); arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->sync_freq); arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->sync_freq); arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); -- cgit From f641aec62c948c7754429136ad176824fbb97238 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:22 +0000 Subject: ASoC: arizona: Calculate OUTDIV first OUTDIV will remain unchanged whilst the rest of the FLL configuration is calculated so do this first. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7b1354ae337b..1f106abf1bb0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1415,6 +1415,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout); + /* Fvco should be over the targt; don't check the upper bound */ + div = ARIZONA_FLL_MIN_OUTDIV; + while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { + div++; + if (div > ARIZONA_FLL_MAX_OUTDIV) + return -EINVAL; + } + target = fll->fout * div / fll->vco_mult; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + /* Fref must be <=13.5MHz */ div = 1; cfg->refdiv = 0; @@ -1429,18 +1441,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Apply the division for our remaining calculations */ Fref /= div; - /* Fvco should be over the targt; don't check the upper bound */ - div = ARIZONA_FLL_MIN_OUTDIV; - while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { - div++; - if (div > ARIZONA_FLL_MAX_OUTDIV) - return -EINVAL; - } - target = fll->fout * div / fll->vco_mult; - cfg->outdiv = div; - - arizona_fll_dbg(fll, "Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { -- cgit From 5a3935c7643966e4172e7a704a48a35f9b4dc668 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:23 +0000 Subject: ASoC: arizona: Calculate FLL gain last No part of the FLL calculation depends on the value determined for the gain but the gain does depend on other values. In preparation for future updates this patch moves the gain to be the last thing calculated. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 1f106abf1bb0..219d1d54f536 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1455,18 +1455,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } - for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { - if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { - cfg->gain = fll_gains[i].gain; - break; - } - } - if (i == ARRAY_SIZE(fll_gains)) { - arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", - Fref); - return -EINVAL; - } - cfg->n = target / (ratio * Fref); if (target % (ratio * Fref)) { @@ -1490,6 +1478,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->lambda >>= 1; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", -- cgit From d0800342bba71e7f11b2a67a29cf00c41ad1a3e4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:25 +0000 Subject: ASoC: arizona: Support new fratio encoding on the wm5110 rev D The reference clock path on newer IP FLLs requires a different configuration, and should avoid integer mode operation. This patch adds support for both the new encoding and updates the calculation. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 130 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 101 insertions(+), 29 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 219d1d54f536..c3884861e8cb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -53,8 +53,10 @@ #define ARIZONA_AIF_RX_ENABLES 0x1A #define ARIZONA_AIF_FORCE_WRITE 0x1B +#define ARIZONA_FLL_VCO_CORNER 141900000 #define ARIZONA_FLL_MAX_FREF 13500000 #define ARIZONA_FLL_MIN_FVCO 90000000 +#define ARIZONA_FLL_MAX_FRATIO 16 #define ARIZONA_FLL_MAX_REFDIV 8 #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 @@ -1406,9 +1408,99 @@ static int arizona_validate_fll(struct arizona_fll *fll, return 0; } +static int arizona_find_fratio(unsigned int Fref, int *fratio) +{ + int i; + + /* Find an appropriate FLL_FRATIO */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + if (fratio) + *fratio = fll_fratios[i].fratio; + return fll_fratios[i].ratio; + } + } + + return -EINVAL; +} + +static int arizona_calc_fratio(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int target, + unsigned int Fref, bool sync) +{ + int init_ratio, ratio; + int refdiv, div; + + /* Fref must be <=13.5MHz, find initial refdiv */ + div = 1; + cfg->refdiv = 0; + while (Fref > ARIZONA_FLL_MAX_FREF) { + div *= 2; + Fref /= 2; + cfg->refdiv++; + + if (div > ARIZONA_FLL_MAX_REFDIV) + return -EINVAL; + } + + /* Find an appropriate FLL_FRATIO */ + init_ratio = arizona_find_fratio(Fref, &cfg->fratio); + if (init_ratio < 0) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return init_ratio; + } + + switch (fll->arizona->type) { + case WM5110: + if (fll->arizona->rev < 3 || sync) + return init_ratio; + break; + default: + return init_ratio; + } + + cfg->fratio = init_ratio - 1; + + /* Adjust FRATIO/refdiv to avoid integer mode if possible */ + refdiv = cfg->refdiv; + + while (div <= ARIZONA_FLL_MAX_REFDIV) { + for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + ratio++) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } + } + + for (ratio = init_ratio - 1; ratio >= 0; ratio--) { + if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < + Fref) + break; + + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } + } + + div *= 2; + Fref /= 2; + refdiv++; + init_ratio = arizona_find_fratio(Fref, NULL); + } + + arizona_fll_warn(fll, "Falling back to integer mode operation\n"); + return cfg->fratio + 1; +} + static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, - unsigned int Fref) + unsigned int Fref, bool sync) { unsigned int target, div, gcd_fll; int i, ratio; @@ -1427,33 +1519,13 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "Fvco=%dHz\n", target); - /* Fref must be <=13.5MHz */ - div = 1; - cfg->refdiv = 0; - while ((Fref / div) > ARIZONA_FLL_MAX_FREF) { - div *= 2; - cfg->refdiv++; - - if (div > ARIZONA_FLL_MAX_REFDIV) - return -EINVAL; - } + /* Find an appropriate FLL_FRATIO and refdiv */ + ratio = arizona_calc_fratio(fll, cfg, target, Fref, sync); + if (ratio < 0) + return ratio; /* Apply the division for our remaining calculations */ - Fref /= div; - - /* Find an appropraite FLL_FRATIO and factor it out of the target */ - for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { - if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { - cfg->fratio = fll_fratios[i].fratio; - ratio = fll_fratios[i].ratio; - break; - } - } - if (i == ARRAY_SIZE(fll_fratios)) { - arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", - Fref); - return -EINVAL; - } + Fref = Fref / (1 << cfg->refdiv); cfg->n = target / (ratio * Fref); @@ -1564,19 +1636,19 @@ static void arizona_enable_fll(struct arizona_fll *fll) */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_calc_fll(fll, &cfg, fll->ref_freq); + arizona_calc_fll(fll, &cfg, fll->ref_freq, false); arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq); + arizona_calc_fll(fll, &cfg, fll->sync_freq, true); arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq); + arizona_calc_fll(fll, &cfg, fll->sync_freq, false); arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); -- cgit From b0ffe9b1ab8705eeffa72feacd791ab6fdd2a06c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 8 Mar 2014 15:47:22 +0100 Subject: ASoC: sam9g20_wm8731: Convert to table based DAPM setup Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. While we are at it also remove the snd_soc_dapm_enable_pin() in the init callback. Pins are enabled by default. Signed-off-by: Lars-Peter Clausen Tested-by: Bo Shen Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f15bff1548f8..174bd546c08b 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -155,25 +155,14 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* Add specific widgets */ - snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets, - ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - /* not connected */ snd_soc_dapm_nc_pin(dapm, "RLINEIN"); snd_soc_dapm_nc_pin(dapm, "LLINEIN"); -#ifdef ENABLE_MIC_INPUT - snd_soc_dapm_enable_pin(dapm, "Int Mic"); -#else - snd_soc_dapm_nc_pin(dapm, "Int Mic"); +#ifndef ENABLE_MIC_INPUT + snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic"); #endif - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - return 0; } @@ -194,6 +183,11 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .dai_link = &at91sam9g20ek_dai, .num_links = 1, .set_bias_level = at91sam9g20ek_set_bias_level, + + .dapm_widgets = at91sam9g20ek_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int at91sam9g20ek_audio_probe(struct platform_device *pdev) -- cgit