November 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for MODX
by Geoffrey D. Bennett 05 Nov '20

05 Nov '20

This patch fixes audio distortion on playback for the Yamaha MODX. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Tested-by: Frank Slotta <frank.slotta(a)posteo.de> --- pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..22fbd1f48b3e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -345,6 +345,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum;

2 3

[PATCH v2 0/6] ASoC: codecs: add support for LPASS Codec macros
by Srinivas Kandagatla 04 Nov '20

04 Nov '20

This patchset adds support for two Codec Macro blocks( WSA and VA) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD Codecs to provide headphone/ear/lineout etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Thanks, srini -Changes since v1: - updated code to fix cppcheck errors suggested by Pierre - fixed various coding style and variable initialization suggested by Pierre. - yaml bindings clock names replaced with constants. Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes .../bindings/sound/qcom,lpass-va-macro.yaml | 67 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 69 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-va-macro.c | 1500 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2472 +++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 15 + 7 files changed, 4135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h -- 2.21.0

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[PATCH v5 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 04 Nov '20

04 Nov '20

Hi Rob, Mark Cc: Sameer These are v5 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v4 -> v5 - move "endpoint" under "port" v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 162 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 620 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

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[PATCH v3] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 04 Nov '20

04 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 ++++++++++++++++ 2 files changed, 143 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..a9cd6ad0bf5a 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,106 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 2: + val |= RT1015_I2S_TX_2CH; + break; + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 16: + val |= RT1015_I2S_CH_TX_LEN_16B; + break; + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + /* This is an assumption that the system sends stereo audio to the amplifier typically. + * And the stereo audio is placed in slot 0/2/4/6 as the starting slot. + * The users could select the channel from L/R/L+R by "Mono LR Select" control. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1075,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

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[PATCH v3 0/7] ASoC: topology: Change to resource managed memory
by Amadeusz Sławiński 04 Nov '20

04 Nov '20

Almost all other allocations in ASoC API are resource managed, the only exception is soc-topology.c. This patchset clean ups few unnecessary functions in preparation for change and then changes to devm_ functions for allocation. Amadeusz Sławiński (6): ASoC: topology: Remove unused functions from topology API ASoC: topology: Remove multistep topology loading ASoC: topology: Unify all device references ASoC: topology: Change allocations to resource managed ASoC: topology: Remove empty functions ASoC: topology: Simplify remove_widget function include/sound/soc-topology.h | 13 +- sound/soc/intel/skylake/skl-topology.c | 9 +- sound/soc/soc-topology.c | 302 +++++-------------------- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/topology.c | 4 +- 5 files changed, 57 insertions(+), 273 deletions(-) -- 2.25.1

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[PATCH] ASoC: mchp-spdiftx: Do not set Validity bit(s)
by Codrin Ciubotariu 04 Nov '20

04 Nov '20

The Validity bits (bit 28) must not be set in order to have the samples valid. Some controllers look for this bit and ignore the samples if it is set. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/mchp-spdiftx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 82c1eecd2528..3bd350afb743 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream, } mchp_spdiftx_channel_status_write(dev); spin_unlock_irqrestore(&ctrl->lock, flags); - mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2; if (dev->gclk_enabled) { clk_disable_unprepare(dev->gclk); -- 2.27.0

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[Feature Request] Add option to change Audio Mixer Source node (0x0c, 0x0d, 0x0e etc)
by GitHub issues - opened 04 Nov '20

04 Nov '20

alsa-project/alsa-tools issue #3 was opened from kevinsmia1939: Hello, I am looking for linux audio tool that can change Audio Mixer Source node, 0x0c to 0x0d for example. HDA Analyzer can do this (right side in red circle of screenshot) https://i.imgur.com/UcELkjU.png But HDA analyzer use python 2 and some other depreciate python modules and since I reinstall my machine (openSUSE TW), I lost the old packages (install old packages manually does not work for some reason). https://www.alsa-project.org/main/index.php/HDA_Analyzer I tried HDAJackRetask, but could not find the option or feature to do it. However, HDAJackRetask is the closest thing to HDA Analyzer, so I think it would make sense to ask for this feature here. If there are already tools that can do this (other than HDA Analyzer) please tell me. Thank you. Issue URL : https://github.com/alsa-project/alsa-tools/issues/3 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Alternative to HDA Analyzer that can change Audio Mixer Source Node?
by andythe_great 04 Nov '20

04 Nov '20

Hello, I need a tool that can change Audio Mixer Source node as seen in this screenshot circle in red.

1 0

[PATCH] sound/core/seq: remove useless function
by Yu Hao 04 Nov '20

04 Nov '20

The function snd_seq_queue_client_termination() is only called from the function seq_free_client1(). The function seq_free_client1() calls the function snd_seq_queue_client_leave() and the function snd_seq_queue_client_termination() together. Because the function snd_seq_queue_client_leave() does all things, so the function snd_seq_queue_client_termination() is a useless function. Signed-off-by: Yu Hao <yuhaobehappy(a)gmail.com> --- sound/core/seq/seq_clientmgr.c | 1 - sound/core/seq/seq_queue.c | 27 --------------------------- sound/core/seq/seq_queue.h | 3 --- 3 files changed, 31 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index cc93157fa950..f9f2fea58b32 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -279,7 +279,6 @@ static int seq_free_client1(struct snd_seq_client *client) snd_seq_delete_all_ports(client); snd_seq_queue_client_leave(client->number); snd_use_lock_sync(&client->use_lock); - snd_seq_queue_client_termination(client->number); if (client->pool) snd_seq_pool_delete(&client->pool); spin_lock_irq(&clients_lock); diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 71a6ea62c3be..13cfc2d47fa7 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -537,33 +537,6 @@ int snd_seq_queue_is_used(int queueid, int client) /*----------------------------------------------------------------*/ -/* notification that client has left the system - - * stop the timer on all queues owned by this client - */ -void snd_seq_queue_client_termination(int client) -{ - unsigned long flags; - int i; - struct snd_seq_queue *q; - bool matched; - - for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) { - if ((q = queueptr(i)) == NULL) - continue; - spin_lock_irqsave(&q->owner_lock, flags); - matched = (q->owner == client); - if (matched) - q->klocked = 1; - spin_unlock_irqrestore(&q->owner_lock, flags); - if (matched) { - if (q->timer->running) - snd_seq_timer_stop(q->timer); - snd_seq_timer_reset(q->timer); - } - queuefree(q); - } -} - /* final stage notification - * remove cells for no longer exist client (for non-owned queue) * or delete this queue (for owned queue) diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 9254c8dbe5e3..1c3a8d3254d9 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -59,9 +59,6 @@ struct snd_seq_queue *snd_seq_queue_alloc(int client, int locked, unsigned int f /* delete queue (destructor) */ int snd_seq_queue_delete(int client, int queueid); -/* notification that client has left the system */ -void snd_seq_queue_client_termination(int client); - /* final stage */ void snd_seq_queue_client_leave(int client); -- 2.17.1

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[PATCH v2] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 04 Nov '20

04 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 ++++++++++++++ sound/soc/codecs/rt1015.c | 20 +++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 4 files changed, 43 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt index fcfd02d8d32f..da9697c80d2b 100644 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -8,10 +8,16 @@ Required properties: - reg : The I2C address of the device. +Optional properties: + +- realtek,realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. Example: rt1015: codec@28 { compatible = "realtek,rt1015"; reg = <0x28>; + realtek,power-up-delay-ms = <50>; }; diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..70a7538d4c89 --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay_ms; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..967193518349 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay_ms = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay_ms); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay-ms", + &rt1015->pdata.power_up_delay_ms); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

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