From c1124c09e1035cabdbc17d4538ae6f922086fec9 Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Wed, 12 Oct 2016 13:53:28 -0500 Subject: ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver. Initial commit of the Cirrus Logic cs35l34 8V boosted class D amplifier. Signed-off-by: Paul Handrigan Signed-off-by: Mark Brown --- include/sound/cs35l34.h | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) create mode 100644 include/sound/cs35l34.h (limited to 'include') diff --git a/include/sound/cs35l34.h b/include/sound/cs35l34.h new file mode 100644 index 000000000000..9c927cffbe46 --- /dev/null +++ b/include/sound/cs35l34.h @@ -0,0 +1,35 @@ +/* + * linux/sound/cs35l34.h -- Platform data for CS35l34 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L34_H +#define __CS35L34_H + +struct cs35l34_platform_data { + /* Set AIF to half drive strength */ + bool aif_half_drv; + /* Digital Soft Ramp Disable */ + bool digsft_disable; + /* Amplifier Invert */ + bool amp_inv; + /* Peak current (mA) */ + unsigned int boost_peak; + /* Boost inductor value (nH) */ + unsigned int boost_ind; + /* Boost Controller Voltage Setting (mV) */ + unsigned int boost_vtge; + /* Gain Change Zero Cross */ + bool gain_zc_disable; + /* SDIN Left/Right Selection */ + unsigned int i2s_sdinloc; + /* TDM Rising Edge */ + bool tdm_rising_edge; +}; + +#endif /* __CS35L34_H */ -- cgit v1.2.3 From d2e3cb85a9c5824fab237ee4577250d0c76969fe Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 29 Sep 2016 14:09:14 +0800 Subject: ASoC: topology: ABI - Define DAI physical PCM data formats Define DAI physical PCM data formats for user space, so users can configure the formats of backends by topology (e.g. the DAI format to set on backend link init). The kernel will also refer to these formats. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 15 ++++++++------- include/uapi/sound/asoc.h | 15 +++++++++++++++ 2 files changed, 23 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 964b7de1a1cc..534aae2f1bc0 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -15,6 +15,7 @@ #include +#include struct snd_pcm_substream; struct snd_soc_dapm_widget; @@ -26,13 +27,13 @@ struct snd_compr_stream; * Describes the physical PCM data formating and clocking. Add new formats * to the end. */ -#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ -#define SND_SOC_DAIFMT_AC97 6 /* AC97 */ -#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ +#define SND_SOC_DAIFMT_I2S SND_SOC_DAI_FORMAT_I2S +#define SND_SOC_DAIFMT_RIGHT_J SND_SOC_DAI_FORMAT_RIGHT_J +#define SND_SOC_DAIFMT_LEFT_J SND_SOC_DAI_FORMAT_LEFT_J +#define SND_SOC_DAIFMT_DSP_A SND_SOC_DAI_FORMAT_DSP_A +#define SND_SOC_DAIFMT_DSP_B SND_SOC_DAI_FORMAT_DSP_B +#define SND_SOC_DAIFMT_AC97 SND_SOC_DAI_FORMAT_AC97 +#define SND_SOC_DAIFMT_PDM SND_SOC_DAI_FORMAT_PDM /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 33d00a4ce656..8d9814b17c46 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -130,6 +130,21 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI physical PCM data formats. + * Add new formats to the end of the list. + */ +#define SND_SOC_DAI_FORMAT_I2S 1 /* I2S mode */ +#define SND_SOC_DAI_FORMAT_RIGHT_J 2 /* Right Justified mode */ +#define SND_SOC_DAI_FORMAT_LEFT_J 3 /* Left Justified mode */ +#define SND_SOC_DAI_FORMAT_DSP_A 4 /* L data MSB after FRM LRC */ +#define SND_SOC_DAI_FORMAT_DSP_B 5 /* L data MSB during FRM LRC */ +#define SND_SOC_DAI_FORMAT_AC97 6 /* AC97 */ +#define SND_SOC_DAI_FORMAT_PDM 7 /* Pulse density modulation */ + +/* left and right justified also known as MSB and LSB respectively */ +#define SND_SOC_DAI_FORMAT_MSB SND_SOC_DAI_FORMAT_LEFT_J +#define SND_SOC_DAI_FORMAT_LSB SND_SOC_DAI_FORMAT_RIGHT_J + /* * Block Header. * This header precedes all object and object arrays below. -- cgit v1.2.3 From 1236fa1e3c29922d201da0926aa8d62427f74814 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Oct 2016 00:44:10 +0000 Subject: ASoC: soc.h: use bit field for playback/capture_only Current snd_soc_dai_link is already using many bit fields. Let's use it for playback_only/capture_only too. We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f1c784e44f6..3d569307f1e2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1025,13 +1025,13 @@ struct snd_soc_dai_link { const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; - /* For unidirectional dai links */ - bool playback_only; - bool capture_only; - /* Mark this pcm with non atomic ops */ bool nonatomic; + /* For unidirectional dai links */ + unsigned int playback_only:1; + unsigned int capture_only:1; + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; -- cgit v1.2.3 From a7df0d3b455cb235926f53e02895d94fccba1e14 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Oct 2016 05:24:36 +0000 Subject: ASoC: bunch up bit field for snd_soc_pcm_runtime We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 3d569307f1e2..d11f1a801ce7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1203,14 +1203,11 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int dev_registered:1; - /* Dynamic PCM BE runtime data */ struct snd_soc_dpcm_runtime dpcm[2]; int fe_compr; long pmdown_time; - unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; @@ -1232,6 +1229,10 @@ struct snd_soc_pcm_runtime { unsigned int num; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ + + /* bit field */ + unsigned int dev_registered:1; + unsigned int pop_wait:1; }; /* mixer control */ -- cgit v1.2.3 From 39b2238e2c4b5a34f53871902458a87729c88ae6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Oct 2016 05:25:19 +0000 Subject: ASoC: bunch up bit field for snd_soc_dai We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 964b7de1a1cc..d0ae96b615ad 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -268,8 +268,9 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + unsigned int probed:1; + unsigned int active; - unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; -- cgit v1.2.3 From 57619b4c9393b8886da90f4ebf29c9f9fe1d07cf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Oct 2016 06:32:44 +0000 Subject: ASoC: remove component from snd_soc_pcm_runtime commit f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") removed its usecase. No one is using it now. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f1c784e44f6..1ed9371ece0d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1219,7 +1219,6 @@ struct snd_soc_pcm_runtime { struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; - struct snd_soc_component *component; /* Only valid for AUX dev rtds */ struct snd_soc_dai **codec_dais; unsigned int num_codecs; -- cgit v1.2.3 From a5461fd6c8f70b3d897f5f76612e890ad47a5b93 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Tue, 25 Oct 2016 19:27:26 +0800 Subject: ASoC: rt5514: Add the DMIC initial delay to wait it ready. Due to the DMIC that needs time to initial after the MCLK is provided, the field of delay time is implemented by the platform data. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- include/sound/rt5514.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 include/sound/rt5514.h (limited to 'include') diff --git a/include/sound/rt5514.h b/include/sound/rt5514.h new file mode 100644 index 000000000000..ef18494769ee --- /dev/null +++ b/include/sound/rt5514.h @@ -0,0 +1,20 @@ +/* + * linux/sound/rt5514.h -- Platform data for RT5514 + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: Oder Chiou + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5514_H +#define __LINUX_SND_RT5514_H + +struct rt5514_platform_data { + unsigned int dmic_init_delay; +}; + +#endif + -- cgit v1.2.3 From c6644119a3f80ea644bde10009d5e1013b5aff29 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Tue, 25 Oct 2016 17:08:39 +0200 Subject: ASoC: Drop SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag Since commit 194c7dea00c68c1b1f8ff26304fa937a006f66dd "ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config" custom DMA channels can be also specified in chan_names[] field of struct snd_dmaengine_pcm_config. This patch removes chan_name field of struct snd_dmaengine_dai_dma_data as it is now unused. Signed-off-by: Sylwester Nawrocki Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 6 ------ 1 file changed, 6 deletions(-) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 67be2445941a..1c8f9e1ef2a5 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -71,7 +71,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. - * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ @@ -81,7 +80,6 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; - const char *chan_name; unsigned int fifo_size; unsigned int flags; }; @@ -107,10 +105,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) -/* - * The PCM streams have custom channel names specified. - */ -#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM -- cgit v1.2.3 From e411b0b5eb9b65257a050eac333d181d6e00e2c6 Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Wed, 2 Nov 2016 15:35:58 +0800 Subject: ASoC: dapm: Support second register for DAPM control updates To support double channel shared controls split across 2 registers, one for each channel, we must be able to update both registers together. Add a second set of register fields to struct snd_soc_dapm_update, and update the DAPM control writeback (put) callbacks to support this. For codecs that use custom events which call into DAPM to do updates, also clear struct snd_soc_dapm_update before using it, so the second set of fields remains clean. Signed-off-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f60d755f7ac6..d5f4677776ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -615,6 +615,10 @@ struct snd_soc_dapm_update { int reg; int mask; int val; + int reg2; + int mask2; + int val2; + bool has_second_set; }; struct snd_soc_dapm_wcache { -- cgit v1.2.3 From 9ee7ef31b5a07cdca88cae023c613e045af935b9 Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Wed, 2 Nov 2016 15:36:00 +0800 Subject: ASoC: dapm: Introduce DAPM_DOUBLE dual channel control type A DAPM_DOUBLE control type can be used for dual channel mixer input selectors / mute controls in one register, possibly toggling both channels together. The control is meant to be shared by 2 widgets, 1 for each channel, such that the mixer control exposed to userspace remains a combined stereo control. Signed-off-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d5f4677776ce..f74ec19687f8 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -272,6 +272,11 @@ struct device; /* dapm kcontrol types */ +#define SOC_DAPM_DOUBLE(xname, reg, lshift, rshift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_DOUBLE_VALUE(reg, lshift, rshift, max, invert, 0) } #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ -- cgit v1.2.3 From 02866eab0f0d88c4b6a68de72022c2b26f0359b5 Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Wed, 2 Nov 2016 15:36:01 +0800 Subject: ASoC: dapm: Introduce DAPM_DOUBLE_R dual channel dual register control type A DAPM_DOUBLE_R control type can be used for dual channel mixer input selectors / mute controls across 2 registers, possibly toggling both channels together. The control is meant to be shared by 2 widgets, 1 for each channel, such that the mixer control exposed to userspace remains a combined stereo control. Signed-off-by: Chen-Yu Tsai Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f74ec19687f8..a466f4bdc835 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -277,6 +277,11 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, lshift, rshift, max, invert, 0) } +#define SOC_DAPM_DOUBLE_R(xname, lreg, rreg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_DOUBLE_R_VALUE(lreg, rreg, shift, max, invert) } #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ -- cgit v1.2.3 From 288b8da7e992f0b86b283f98e92885781ffdcaee Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:03:17 +0800 Subject: ASoC: topology: Support topology file of ABI v4 Users start to use topology ABI from v4. ABI v5 updated existing manifest and PCM elements. Two previous patches can support these ABI updates in a backward compatible way. So if the topology file from user space is generated by ABI v4, kernel will no longer quit but continue parsing. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 8d9814b17c46..69cae7198b18 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -83,7 +83,8 @@ #define SND_SOC_TPLG_NUM_TEXTS 16 /* ABI version */ -#define SND_SOC_TPLG_ABI_VERSION 0x5 +#define SND_SOC_TPLG_ABI_VERSION 0x5 /* current version */ +#define SND_SOC_TPLG_ABI_VERSION_MIN 0x4 /* oldest version supported */ /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 -- cgit v1.2.3 From 717a8e7235377fa50f074c407cc5d0486aa15aa2 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:03:34 +0800 Subject: ASoC: topology: ABI - Add flags and private data to PCM This is the remaining update to PCM ABI object of version 5. The flags will be applied to FE (Front End) links and can also be used by physical links. The private data is reserved for future extension, so offset update will add the private data size. Now user space is using ABI v4, and the previous patch "ASoC: topology: make PCM backward compatible from ABI v4" can assure the backward compatibility. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69cae7198b18..aeb408241bc3 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -146,6 +146,11 @@ #define SND_SOC_DAI_FORMAT_MSB SND_SOC_DAI_FORMAT_LEFT_J #define SND_SOC_DAI_FORMAT_LSB SND_SOC_DAI_FORMAT_RIGHT_J +/* DAI link flags */ +#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_RATES (1 << 0) +#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) +#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) + /* * Block Header. * This header precedes all object and object arrays below. @@ -456,6 +461,9 @@ struct snd_soc_tplg_pcm { struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */ __le32 num_streams; /* number of streams */ struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */ + __le32 flag_mask; /* bitmask of flags to configure */ + __le32 flags; /* SND_SOC_TPLG_LNK_FLGBIT_* flag value */ + struct snd_soc_tplg_private priv; } __attribute__((packed)); -- cgit v1.2.3 From 8a0cb2360ddb941e0a2fbe33d400d2985e4f2fff Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 3 Nov 2016 17:07:18 +0530 Subject: ASoC: Intel: Skylake: Add support for LPMode For D0i3, we need to tell DSP to run the pipelines in LP mode. This information is kept in topology and passed to driver as an attribute for pipe. So add a new tuple for lpmode and program the pipe based on value set. Signed-off-by: Jayachandran B Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/snd_sst_tokens.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index 1ee2e943d66a..f4b8b34de519 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -157,6 +157,8 @@ * * %SKL_TKN_STR_LIB_NAME: Specifies the library name * + * %SKL_TKN_U32_PMODE: Specifies the power mode for pipe + * * module_id and loadable flags dont have tokens as these values will be * read from the DSP FW manifest */ @@ -208,7 +210,8 @@ enum SKL_TKNS { SKL_TKN_U32_PROC_DOMAIN, SKL_TKN_U32_LIB_COUNT, SKL_TKN_STR_LIB_NAME, - SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME, + SKL_TKN_U32_PMODE, + SKL_TKN_MAX = SKL_TKN_U32_PMODE, }; #endif -- cgit v1.2.3 From 6bd9dcf339ebb292fa149ee0e52ea3c9d9822553 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 3 Nov 2016 17:07:19 +0530 Subject: ASoC: Intel: Skylake: Add support for specifying D0i3 configuration Not all use cases can support Doi3. Only certain use cases like hot word detection, deep buffering can support D0i3 based on resource requirement. So, pass the D0i3 capability for the FE/BE copier using topology. This will be used to take a decision for D0i3 mode entry/exit. Signed-off-by: Jayachandran B Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/uapi/sound/snd_sst_tokens.h | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h index f4b8b34de519..93392bedcc58 100644 --- a/include/uapi/sound/snd_sst_tokens.h +++ b/include/uapi/sound/snd_sst_tokens.h @@ -159,6 +159,8 @@ * * %SKL_TKN_U32_PMODE: Specifies the power mode for pipe * + * %SKL_TKL_U32_D0I3_CAPS: Specifies the D0i3 capability for module + * * module_id and loadable flags dont have tokens as these values will be * read from the DSP FW manifest */ @@ -211,7 +213,8 @@ enum SKL_TKNS { SKL_TKN_U32_LIB_COUNT, SKL_TKN_STR_LIB_NAME, SKL_TKN_U32_PMODE, - SKL_TKN_MAX = SKL_TKN_U32_PMODE, + SKL_TKL_U32_D0I3_CAPS, + SKL_TKN_MAX = SKL_TKL_U32_D0I3_CAPS, }; #endif -- cgit v1.2.3 From 676c6b5208f7ce6ccc7933ab210f24358ca99a7a Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:03:52 +0800 Subject: ASoC: topology: ABI - Update physical DAI link configuration for version 5 The following fields are added to physical link configuration struct (snd_soc_tplg_link_config) in ABI v5: - name and stream name Topology will use them to find an existing physical link and configure it. - HW configurations Define the types and ABI struct for runtime supported hardware configs of physical DAI links, e.g. audio hardware formats. The default HW config ID will help topology to find the DAI format to set on init. Topology provides this as a fallback if such HW settings are not available in ACPI or device tree, to avoid hard code in drivers. It's only for config items that can be programmed by SW or FW, not for physical things like link connections or GPIO used for HP etc. - flags and private data The flags will be used to configure an existing physical DAI link. The private data is reserved for future extension. NOTE: Current kernel has no support for physical links. A later patch will add support for configuring physical links and make the support backward compatible for ABI v4. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 46 ++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 44 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index aeb408241bc3..ed28ebc392d1 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -39,6 +39,11 @@ */ #define SND_SOC_TPLG_STREAM_CONFIG_MAX 8 +/* + * Maximum number of physical link's hardware configs + */ +#define SND_SOC_TPLG_HW_CONFIG_MAX 8 + /* individual kcontrol info types - can be mixed with other types */ #define SND_SOC_TPLG_CTL_VOLSW 1 #define SND_SOC_TPLG_CTL_VOLSW_SX 2 @@ -294,6 +299,35 @@ struct snd_soc_tplg_stream { __le32 channels; /* channels */ } __attribute__((packed)); + +/* + * Describes a physical link's runtime supported hardware config, + * i.e. hardware audio formats. + */ +struct snd_soc_tplg_hw_config { + __le32 size; /* in bytes of this structure */ + __le32 id; /* unique ID - - used to match */ + __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ + __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ + __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ + __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ + __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 mclk_direction; /* 0 for input, 1 for output */ + __le16 reserved; /* for 32bit alignment */ + __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ + __le32 bclk_rate; /* BCLK freqency in Hz */ + __le32 fsync_rate; /* frame clock in Hz */ + __le32 tdm_slots; /* number of TDM slots in use */ + __le32 tdm_slot_width; /* width in bits for each slot */ + __le32 tx_slots; /* bit mask for active Tx slots */ + __le32 rx_slots; /* bit mask for active Rx slots */ + __le32 tx_channels; /* number of Tx channels */ + __le32 tx_chanmap[SND_SOC_TPLG_MAX_CHAN]; /* array of slot number */ + __le32 rx_channels; /* number of Rx channels */ + __le32 rx_chanmap[SND_SOC_TPLG_MAX_CHAN]; /* array of slot number */ +} __attribute__((packed)); + /* * Manifest. List totals for each payload type. Not used in parsing, but will * be passed to the component driver before any other objects in order for any @@ -468,9 +502,9 @@ struct snd_soc_tplg_pcm { /* - * Describes the BE or CC link runtime supported configs or params + * Describes the physical link runtime supported configs or params * - * File block representation for BE/CC link config :- + * File block representation for physical link config :- * +-----------------------------------+-----+ * | struct snd_soc_tplg_hdr | 1 | * +-----------------------------------+-----+ @@ -480,8 +514,16 @@ struct snd_soc_tplg_pcm { struct snd_soc_tplg_link_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - used to match */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */ + char stream_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* stream name - used to match */ struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */ __le32 num_streams; /* number of streams */ + struct snd_soc_tplg_hw_config hw_config[SND_SOC_TPLG_HW_CONFIG_MAX]; /* hw configs */ + __le32 num_hw_configs; /* number of hw configs */ + __le32 default_hw_config_id; /* default hw config ID for init */ + __le32 flag_mask; /* bitmask of flags to configure */ + __le32 flags; /* SND_SOC_TPLG_LNK_FLGBIT_* flag value */ + struct snd_soc_tplg_private priv; } __attribute__((packed)); /* -- cgit v1.2.3 From 17fb175520e5497d71351aa66a125324fcb625a7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:04:12 +0800 Subject: ASoC: Define API to find a dai link Define the API to find an existing DAI link of the soc card by matching the ID, name and stream name. Some cards may use unique ID for each DAI link, so matching ID is enough, and name or stream name are not necessary. But user need to specify name or stream name as well if not sure whether link ID is unique since most cards use 0 as the default link ID. Topology can use this API to find an existing BE link and configure it. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f1c784e44f6..c3a38ee2b006 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1671,6 +1671,9 @@ int snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); void snd_soc_remove_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card, + int id, const char *name, + const char *stream_name); int snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv); -- cgit v1.2.3 From 3fbf793510c7628248a965972112fab958e6e3cf Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:05:01 +0800 Subject: ASoC: topology: ABI - Rename struct and type for physical DAIs Rename the ABI struct and type because they are for configuring physical DAIs, not only backend DAIs since users may not need DPCM: - Rename struct snd_soc_tplg_be_dai to snd_soc_tplg_dai. - Rename type SND_SOC_TPLG_TYPE_BE_DAI to SND_SOC_TPLG_TYPE_DAI. This code refactoring is backward compatible because: - Both layout of the struct and type value has no change. Kernel can find the same type value and map to same data layout. - This struct is not in ABI v4 at all. Now the user space uses ABI v4. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index ed28ebc392d1..3c3fcc86b9f4 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -111,8 +111,8 @@ #define SND_SOC_TPLG_TYPE_CODEC_LINK 9 #define SND_SOC_TPLG_TYPE_BACKEND_LINK 10 #define SND_SOC_TPLG_TYPE_PDATA 11 -#define SND_SOC_TPLG_TYPE_BE_DAI 12 -#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_BE_DAI +#define SND_SOC_TPLG_TYPE_DAI 12 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_DAI /* vendor block IDs - please add new vendor types to end */ #define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 @@ -131,7 +131,7 @@ #define SND_SOC_TPLG_TUPLE_TYPE_WORD 4 #define SND_SOC_TPLG_TUPLE_TYPE_SHORT 5 -/* BE DAI flags */ +/* DAI flags */ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES (1 << 0) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) @@ -527,16 +527,17 @@ struct snd_soc_tplg_link_config { } __attribute__((packed)); /* - * Describes SW/FW specific features of BE DAI. + * Describes SW/FW specific features of physical DAI. + * It can be used to configure backend DAIs for DPCM. * - * File block representation for BE DAI :- + * File block representation for physical DAI :- * +-----------------------------------+-----+ * | struct snd_soc_tplg_hdr | 1 | * +-----------------------------------+-----+ - * | struct snd_soc_tplg_be_dai | N | + * | struct snd_soc_tplg_dai | N | * +-----------------------------------+-----+ */ -struct snd_soc_tplg_be_dai { +struct snd_soc_tplg_dai { __le32 size; /* in bytes of this structure */ char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */ __le32 dai_id; /* unique ID - used to match */ -- cgit v1.2.3 From 6ff67ccafdf4c782489de1ccc47e1ec8d8480b63 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 3 Nov 2016 01:05:32 +0800 Subject: ASoC: topology: ABI - Add voice wake up flag for DAI links Add a new flag bit SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP to link flags. If a link is used for voice wake up, users can set this flag bit and topology will set the link's 'ignore_suspend' to true. This ABI update is backward compatible. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 3c3fcc86b9f4..6a4280c6e860 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -155,6 +155,7 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_RATES (1 << 0) #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +#define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) /* * Block Header. -- cgit v1.2.3 From 2c394ca79604b404fe60218670ab301ecb758b34 Mon Sep 17 00:00:00 2001 From: James Schulman Date: Mon, 7 Nov 2016 14:38:37 -0600 Subject: ASoC: Add support for CS42L42 codec Add support for Cirrus Logic CS42L42 codec. SoundWire support is not enabled. Features support for I2C control and I2S audio. Signed-off-by: James Schulman Signed-off-by: Mark Brown --- include/dt-bindings/sound/cs42l42.h | 73 +++++++++++++++++++++++++++++++++++++ 1 file changed, 73 insertions(+) create mode 100644 include/dt-bindings/sound/cs42l42.h (limited to 'include') diff --git a/include/dt-bindings/sound/cs42l42.h b/include/dt-bindings/sound/cs42l42.h new file mode 100644 index 000000000000..399a123aed58 --- /dev/null +++ b/include/dt-bindings/sound/cs42l42.h @@ -0,0 +1,73 @@ +/* + * cs42l42.h -- CS42L42 ALSA SoC audio driver DT bindings header + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: James Schulman + * Author: Brian Austin + * Author: Michael White + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __DT_CS42L42_H +#define __DT_CS42L42_H + +/* HPOUT Load Capacity */ +#define CS42L42_HPOUT_LOAD_1NF 0 +#define CS42L42_HPOUT_LOAD_10NF 1 + +/* HPOUT Clamp to GND Overide */ +#define CS42L42_HPOUT_CLAMP_EN 0 +#define CS42L42_HPOUT_CLAMP_DIS 1 + +/* Tip Sense Inversion */ +#define CS42L42_TS_INV_DIS 0 +#define CS42L42_TS_INV_EN 1 + +/* Tip Sense Debounce */ +#define CS42L42_TS_DBNCE_0 0 +#define CS42L42_TS_DBNCE_125 1 +#define CS42L42_TS_DBNCE_250 2 +#define CS42L42_TS_DBNCE_500 3 +#define CS42L42_TS_DBNCE_750 4 +#define CS42L42_TS_DBNCE_1000 5 +#define CS42L42_TS_DBNCE_1250 6 +#define CS42L42_TS_DBNCE_1500 7 + +/* Button Press Software Debounce Times */ +#define CS42L42_BTN_DET_INIT_DBNCE_MIN 0 +#define CS42L42_BTN_DET_INIT_DBNCE_DEFAULT 100 +#define CS42L42_BTN_DET_INIT_DBNCE_MAX 200 + +#define CS42L42_BTN_DET_EVENT_DBNCE_MIN 0 +#define CS42L42_BTN_DET_EVENT_DBNCE_DEFAULT 10 +#define CS42L42_BTN_DET_EVENT_DBNCE_MAX 20 + +/* Button Detect Level Sensitivities */ +#define CS42L42_NUM_BIASES 4 + +#define CS42L42_HS_DET_LEVEL_15 0x0F +#define CS42L42_HS_DET_LEVEL_8 0x08 +#define CS42L42_HS_DET_LEVEL_4 0x04 +#define CS42L42_HS_DET_LEVEL_1 0x01 + +#define CS42L42_HS_DET_LEVEL_MIN 0 +#define CS42L42_HS_DET_LEVEL_MAX 0x3F + +/* HS Bias Ramp Rate */ + +#define CS42L42_HSBIAS_RAMP_FAST_RISE_SLOW_FALL 0 +#define CS42L42_HSBIAS_RAMP_FAST 1 +#define CS42L42_HSBIAS_RAMP_SLOW 2 +#define CS42L42_HSBIAS_RAMP_SLOWEST 3 + +#define CS42L42_HSBIAS_RAMP_TIME0 10 +#define CS42L42_HSBIAS_RAMP_TIME1 40 +#define CS42L42_HSBIAS_RAMP_TIME2 90 +#define CS42L42_HSBIAS_RAMP_TIME3 170 + +#endif /* __DT_CS42L42_H */ -- cgit v1.2.3 From 17e593e32cebd0d79c7317d35b7e3b022fdfcefa Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Sat, 5 Nov 2016 08:42:24 +0800 Subject: ASoC: topology: ABI - Rename be_dai_elems to dai_elems in manifest User space uses this field to count physical DAIs, not only BE DAIs since users may not use DPCM. So we rename this field from be_dai_elems to dai_elems. This change is backward compatible, because it does not change the layout of the struct or data type. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6a4280c6e860..721dd941f171 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -348,7 +348,7 @@ struct snd_soc_tplg_manifest { __le32 graph_elems; /* number of graph elements */ __le32 pcm_elems; /* number of PCM elements */ __le32 dai_link_elems; /* number of DAI link elements */ - __le32 be_dai_elems; /* number of BE DAI elements */ + __le32 dai_elems; /* number of physical DAI elements */ __le32 reserved[20]; /* reserved for new ABI element types */ struct snd_soc_tplg_private priv; } __attribute__((packed)); -- cgit v1.2.3 From 7364c8dc255232db33bcd1c5b19eb8f34cf6108a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Nov 2016 01:18:35 +0000 Subject: ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing It is assuming that the card related information is located on "card" node, but graph case doesn't have it. This patch adds node parameter to adjust for graph support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f1c784e44f6..148bf553d598 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1655,8 +1655,13 @@ void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, struct snd_soc_codec_conf *codec_conf, struct device_node *of_node, const char *propname); -int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, - const char *propname); + +#define snd_soc_of_parse_audio_routing(card, propname) \ + snd_soc_of_parse_audio_routing_from_node(card, NULL, propname) +int snd_soc_of_parse_audio_routing_from_node(struct snd_soc_card *card, + struct device_node *np, + const char *propname); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, -- cgit v1.2.3 From 1ad8ec535b997ed36c0f32c2616206725258dd30 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Nov 2016 01:19:28 +0000 Subject: ASoC: soc-core: snd_soc_get_dai_name() become non static snd_soc_get_dai_name() is used from snd_soc_of_get_dai_name(), and it is assuming that DT is using "sound-dai" / "#sound-dai-cells". But graph base DT is using "remote-endpoint". This patch makes snd_soc_get_dai_name() non static for graph support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 148bf553d598..6dffa9540a30 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1666,6 +1666,8 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix, struct device_node **bitclkmaster, struct device_node **framemaster); +int snd_soc_get_dai_name(struct of_phandle_args *args, + const char **dai_name); int snd_soc_of_get_dai_name(struct device_node *of_node, const char **dai_name); int snd_soc_of_get_dai_link_codecs(struct device *dev, -- cgit v1.2.3 From b6defcca0a604129155ae472b116a2e1688d8995 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Nov 2016 01:19:03 +0000 Subject: ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix It is assuming that the card related information is located on "card" node, but graph case doesn't have it. This patch adds node parameter to adjust for graph support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6dffa9540a30..346223b38e0d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1651,7 +1651,11 @@ int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width); -void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, +#define snd_soc_of_parse_audio_prefix(card, codec_conf, of_node, propname) \ + snd_soc_of_parse_audio_prefix_from_node(card, NULL, codec_conf, \ + of_node, propname) +void snd_soc_of_parse_audio_prefix_from_node(struct snd_soc_card *card, + struct device_node *np, struct snd_soc_codec_conf *codec_conf, struct device_node *of_node, const char *propname); -- cgit v1.2.3 From 8f5ebb1bee15b5720741a98414767bb86f6c2b23 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Nov 2016 01:17:27 +0000 Subject: ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name It is assuming that the card related information is located on "card" node, but graph case doesn't have it. This patch adds node parameter to adjust for graph support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 346223b38e0d..8457aab9eab0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1642,8 +1642,11 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( int snd_soc_util_init(void); void snd_soc_util_exit(void); -int snd_soc_of_parse_card_name(struct snd_soc_card *card, - const char *propname); +#define snd_soc_of_parse_card_name(card, propname) \ + snd_soc_of_parse_card_name_from_node(card, NULL, propname) +int snd_soc_of_parse_card_name_from_node(struct snd_soc_card *card, + struct device_node *np, + const char *propname); int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_tdm_slot(struct device_node *np, -- cgit v1.2.3 From 1ef5bcd57be5c8b31286b7b47828064be25f266b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 11 Nov 2016 01:18:08 +0000 Subject: ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets It is assuming that the card related information is located on "card" node, but graph case doesn't have it. This patch adds node parameter to adjust for graph support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 8457aab9eab0..77b01c45ee8d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1647,8 +1647,12 @@ void snd_soc_util_exit(void); int snd_soc_of_parse_card_name_from_node(struct snd_soc_card *card, struct device_node *np, const char *propname); -int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, - const char *propname); +#define snd_soc_of_parse_audio_simple_widgets(card, propname)\ + snd_soc_of_parse_audio_simple_widgets_from_node(card, NULL, propname) +int snd_soc_of_parse_audio_simple_widgets_from_node(struct snd_soc_card *card, + struct device_node *np, + const char *propname); + int snd_soc_of_parse_tdm_slot(struct device_node *np, unsigned int *tx_mask, unsigned int *rx_mask, -- cgit v1.2.3 From 2e622ae41e653cd71a6d3bc5a4685b01eacbee8d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Sun, 13 Nov 2016 12:10:02 +0530 Subject: ASoC: compress: Add support for compress dai ops ASoC Compress ops have only platform ops and no DAI ops unlike PCM device where we have both platform ops as well as DAI ops. So add compress dai ops and add this new structure to the ASoC core to make compressed devices a first class ASoC citizen Again like PCM ops, drivers are free to implement either or both of these ops based on device needs. Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 964b7de1a1cc..756ee1b78ffc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -207,6 +207,30 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); }; +struct snd_soc_cdai_ops { + /* + * for compress ops + */ + int (*startup)(struct snd_compr_stream *, + struct snd_soc_dai *); + int (*shutdown)(struct snd_compr_stream *, + struct snd_soc_dai *); + int (*set_params)(struct snd_compr_stream *, + struct snd_compr_params *, struct snd_soc_dai *); + int (*get_params)(struct snd_compr_stream *, + struct snd_codec *, struct snd_soc_dai *); + int (*set_metadata)(struct snd_compr_stream *, + struct snd_compr_metadata *, struct snd_soc_dai *); + int (*get_metadata)(struct snd_compr_stream *, + struct snd_compr_metadata *, struct snd_soc_dai *); + int (*trigger)(struct snd_compr_stream *, int, + struct snd_soc_dai *); + int (*pointer)(struct snd_compr_stream *, + struct snd_compr_tstamp *, struct snd_soc_dai *); + int (*ack)(struct snd_compr_stream *, size_t, + struct snd_soc_dai *); +}; + /* * Digital Audio Interface Driver. * @@ -236,6 +260,7 @@ struct snd_soc_dai_driver { /* ops */ const struct snd_soc_dai_ops *ops; + const struct snd_soc_cdai_ops *cops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; -- cgit v1.2.3 From 33ada14a26c8f174dac8765f4236ca66c64ae5be Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 14 Nov 2016 11:00:10 +0800 Subject: ASoC: add rt5665 codec driver This is the initial codec driver for rt5665. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5665.h | 47 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) create mode 100755 include/sound/rt5665.h (limited to 'include') diff --git a/include/sound/rt5665.h b/include/sound/rt5665.h new file mode 100755 index 000000000000..963229e71dc7 --- /dev/null +++ b/include/sound/rt5665.h @@ -0,0 +1,47 @@ +/* + * linux/sound/rt5665.h -- Platform data for RT5665 + * + * Copyright 2016 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5665_H +#define __LINUX_SND_RT5665_H + +enum rt5665_dmic1_data_pin { + RT5665_DMIC1_NULL, + RT5665_DMIC1_DATA_GPIO4, + RT5665_DMIC1_DATA_IN2N, +}; + +enum rt5665_dmic2_data_pin { + RT5665_DMIC2_NULL, + RT5665_DMIC2_DATA_GPIO5, + RT5665_DMIC2_DATA_IN2P, +}; + +enum rt5665_jd_src { + RT5665_JD_NULL, + RT5665_JD1, +}; + +struct rt5665_platform_data { + bool in1_diff; + bool in2_diff; + bool in3_diff; + bool in4_diff; + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5665_dmic1_data_pin dmic1_data_pin; + enum rt5665_dmic2_data_pin dmic2_data_pin; + enum rt5665_jd_src jd_src; + + unsigned int sar_hs_type; +}; + +#endif + -- cgit v1.2.3 From f84551e45fd72a248ee69a5b95383b0334a2faac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2016 22:22:27 +0100 Subject: ALSA: compress: Fix kernel-doc warnings Some fields in struct snd_compr have no corresponding comments, and the kernel-doc complains like: ./include/sound/compress_driver.h:162: warning: No description found for parameter 'id[64]' ./include/sound/compress_driver.h:162: warning: No description found for parameter 'proc_root' ./include/sound/compress_driver.h:162: warning: No description found for parameter 'proc_info_entry' Actually all these are internal elements, just put "private:" comment so that they will be ignored. Acked-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index cee8c00f3d3e..9924bc9cbc7c 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -155,6 +155,7 @@ struct snd_compr { struct mutex lock; int device; #ifdef CONFIG_SND_VERBOSE_PROCFS + /* private: */ char id[64]; struct snd_info_entry *proc_root; struct snd_info_entry *proc_info_entry; -- cgit v1.2.3 From 43e575fabbaa1cc9e25dd0bb31eb6657ac7092ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Nov 2016 22:25:33 +0100 Subject: ALSA: core: Fix kernel-doc warnings Several lines in sound/core.h get the kernel-doc warnings like ./include/sound/core.h:323: warning: No description found for parameter '...' where we use define like foo(x, args...) and "args" isn't mentioned in the comments. As an easy workaround, use simple __VA_ARGS__ for VLA in macros. Signed-off-by: Takashi Iwai --- include/sound/core.h | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index 31079ea5e484..f7d8c10c4c45 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -308,8 +308,8 @@ __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, const char *format, ...); #else -#define __snd_printk(level, file, line, format, args...) \ - printk(format, ##args) +#define __snd_printk(level, file, line, format, ...) \ + printk(format, ##__VA_ARGS__) #endif /** @@ -319,8 +319,8 @@ void __snd_printk(unsigned int level, const char *file, int line, * Works like printk() but prints the file and the line of the caller * when configured with CONFIG_SND_VERBOSE_PRINTK. */ -#define snd_printk(fmt, args...) \ - __snd_printk(0, __FILE__, __LINE__, fmt, ##args) +#define snd_printk(fmt, ...) \ + __snd_printk(0, __FILE__, __LINE__, fmt, ##__VA_ARGS__) #ifdef CONFIG_SND_DEBUG /** @@ -330,10 +330,10 @@ void __snd_printk(unsigned int level, const char *file, int line, * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG is not set. */ -#define snd_printd(fmt, args...) \ - __snd_printk(1, __FILE__, __LINE__, fmt, ##args) -#define _snd_printd(level, fmt, args...) \ - __snd_printk(level, __FILE__, __LINE__, fmt, ##args) +#define snd_printd(fmt, ...) \ + __snd_printk(1, __FILE__, __LINE__, fmt, ##__VA_ARGS__) +#define _snd_printd(level, fmt, ...) \ + __snd_printk(level, __FILE__, __LINE__, fmt, ##__VA_ARGS__) /** * snd_BUG - give a BUG warning message and stack trace @@ -383,8 +383,8 @@ static inline bool snd_printd_ratelimit(void) { return false; } * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. */ -#define snd_printdd(format, args...) \ - __snd_printk(2, __FILE__, __LINE__, format, ##args) +#define snd_printdd(format, ...) \ + __snd_printk(2, __FILE__, __LINE__, format, ##__VA_ARGS__) #else __printf(1, 2) static inline void snd_printdd(const char *format, ...) {} -- cgit v1.2.3 From aeaa6203b6c41d9add8932dbd95bc741839054d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 2 Nov 2016 18:38:39 +0100 Subject: ALSA: emu10k1: Use workqueue instead of kthread for emu1010 fw polling This patch is a cleanup of EMU1010 dock probing code in emu10k1 driver to use work instead of kthread in a loop. The work is lighter and easier to control than kthread, in general. Instead of a loop with the explicit sleep, we do simply delayed-schedule the work. At suspend/resume callbacks, the work is canceled and restarted, respectively. Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 5bd134651f5e..4f42affe777c 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1688,7 +1688,8 @@ struct snd_emu1010 { unsigned int internal_clock; /* 44100 or 48000 */ unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ - struct task_struct *firmware_thread; + struct delayed_work firmware_work; + u32 last_reg; }; struct snd_emu10k1 { -- cgit v1.2.3 From 1b4d9c22191583ef1fb7433417b2ceb2a608d887 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 29 Nov 2016 15:44:38 +0000 Subject: ASoC: core: Add component pin control functions It's often the case that a codec driver will need to control its own pins. However, if a name_prefix has been applied to this codec it must be included in the name passed to any of the snd_soc_dapm_x_pin() functions. The behaviour of the existing pin control functions is reasonable, since you may want to search for a fully-specified name within the scope of an entire card. This means that we can't apply the prefix in these functions because it will break card-scope searches. Constructing a prefixed string "manually" in codec drivers leads to a lot of repetition of the same code. To make this tidier in codec drivers this patch adds a new set of equivalent functions that take a struct snd_soc_component instead of a dapm context and automatically add the component's name_prefix to the given name. This makes it a simple change in codec drivers to be prefix-safe. The new functions are not quite trivial enough to be inlines and the compiler won't be able to compile-away any part of them. Although it looks somewhat inefficient to have to allocate a temporary buffer and combine strings, the current design of the widget list doesn't lend itself to a more optimized implementation - it's a single list of all widgets on a card and is searched linearly for a matching string. As pin state changes are generally low-frequency events it's unlikely to be a significant issue - at least not enough to rewrite the widget list handling just for this. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- include/sound/soc.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f1c784e44f6..a26c651cb1ee 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1697,4 +1697,24 @@ static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm) mutex_unlock(&dapm->card->dapm_mutex); } +int snd_soc_component_enable_pin(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_disable_pin(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_nc_pin(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_get_pin_status(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_force_enable_pin(struct snd_soc_component *component, + const char *pin); +int snd_soc_component_force_enable_pin_unlocked( + struct snd_soc_component *component, + const char *pin); + #endif -- cgit v1.2.3 From d9fc40639dc1b87915426d2ef489a591b0fd650b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Nov 2016 06:22:36 +0000 Subject: ASoC: core: replace codec_dev_list to component_dev_list on Card Current Card has Codec list (= codec_dev_list), but Codec will be removed in the future. Because of this reason, this patch adds new Component list in Card, and replace Codec list. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 1ed9371ece0d..b5544e2287b8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -810,6 +810,7 @@ struct snd_soc_component { struct list_head list; struct list_head list_aux; /* for auxiliary component of the card */ + struct list_head card_list; struct snd_soc_dai_driver *dai_drv; int num_dai; @@ -868,7 +869,6 @@ struct snd_soc_codec { const struct snd_soc_codec_driver *driver; struct list_head list; - struct list_head card_list; /* runtime */ unsigned int cache_bypass:1; /* Suppress access to the cache */ @@ -1170,7 +1170,7 @@ struct snd_soc_card { struct work_struct deferred_resume_work; /* lists of probed devices belonging to this card */ - struct list_head codec_dev_list; + struct list_head component_dev_list; struct list_head widgets; struct list_head paths; @@ -1540,11 +1540,11 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) { - INIT_LIST_HEAD(&card->codec_dev_list); INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); + INIT_LIST_HEAD(&card->component_dev_list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) -- cgit v1.2.3 From 1a653aa44725668590b36bbe2d7fe4736a69f055 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Nov 2016 06:22:55 +0000 Subject: ASoC: core: replace aux_comp_list to component_dev_list Now, Card has component_dev_list, we can replace aux_comp_list to component_dev_list with new auxiliary flags Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index b5544e2287b8..0bd57b77d010 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -807,9 +807,9 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; + unsigned int auxiliary:1; /* for auxiliary component of the card */ struct list_head list; - struct list_head list_aux; /* for auxiliary component of the card */ struct list_head card_list; struct snd_soc_dai_driver *dai_drv; @@ -1148,7 +1148,6 @@ struct snd_soc_card { */ struct snd_soc_aux_dev *aux_dev; int num_aux_devs; - struct list_head aux_comp_list; const struct snd_kcontrol_new *controls; int num_controls; @@ -1543,7 +1542,6 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->widgets); INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); - INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); } -- cgit v1.2.3 From 9178feb4538e055bf22be44c38b90cc31d2baf99 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 30 Nov 2016 06:23:13 +0000 Subject: ASoC: add Component level suspend/resume In current ALSA SoC, Codec only has suspend/resume feature, but it should be supported on Component level. This patch adds it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0bd57b77d010..13cc64e5bb77 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -782,6 +782,8 @@ struct snd_soc_component_driver { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); /* DT */ int (*of_xlate_dai_name)(struct snd_soc_component *component, @@ -808,6 +810,7 @@ struct snd_soc_component { unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ unsigned int registered_as_component:1; unsigned int auxiliary:1; /* for auxiliary component of the card */ + unsigned int suspended:1; /* is in suspend PM state */ struct list_head list; struct list_head card_list; @@ -853,6 +856,8 @@ struct snd_soc_component { int (*probe)(struct snd_soc_component *); void (*remove)(struct snd_soc_component *); + int (*suspend)(struct snd_soc_component *); + int (*resume)(struct snd_soc_component *); /* machine specific init */ int (*init)(struct snd_soc_component *component); @@ -872,7 +877,6 @@ struct snd_soc_codec { /* runtime */ unsigned int cache_bypass:1; /* Suppress access to the cache */ - unsigned int suspended:1; /* Codec is in suspend PM state */ unsigned int cache_init:1; /* codec cache has been initialized */ /* codec IO */ -- cgit v1.2.3 From 29a43aa9b4b609eba73e76c5fd92d435fc6e35ce Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 2 Dec 2016 05:27:30 +0000 Subject: ASoC: simple_card_utils: tidyup file comment/define simple_card_utils was created as simple_card_core in 1st prototype, and current code still have it. Let's tidyup Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index fd6412551145..64e90ca9ad32 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -1,5 +1,5 @@ /* - * simple_card_core.h + * simple_card_utils.h * * Copyright (c) 2016 Kuninori Morimoto * @@ -7,8 +7,8 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#ifndef __SIMPLE_CARD_CORE_H -#define __SIMPLE_CARD_CORE_H +#ifndef __SIMPLE_CARD_UTILS_H +#define __SIMPLE_CARD_UTILS_H #include @@ -68,4 +68,4 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link, int asoc_simple_card_clean_reference(struct snd_soc_card *card); -#endif /* __SIMPLE_CARD_CORE_H */ +#endif /* __SIMPLE_CARD_UTILS_H */ -- cgit v1.2.3 From eea3dd4f1247aa8654194fb19ade22c94c42e41a Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Fri, 25 Nov 2016 16:09:17 +0800 Subject: ASoC: topology: Only free TLV for volume mixers of a widget This patch will check the type of embedded controls for a widget, and only free the TLV of volume mixers. Bytes controls don't have TLV. Just free the private value which is used as struct soc_mixer_control for volume mixers or soc_bytes_ext for bytes controls. No need to cast to these types before freeing it. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/sound/soc-topology.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index b897b9d63161..f9cc7b9271ac 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -53,7 +53,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ + unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ -- cgit v1.2.3