July 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix a typo in the name of a constant
by Christophe JAILLET 23 Jul '19

23 Jul '19

There is a typo in PCM1368A_MAX_SYSCLK, it should be PCM3168A_MAX_SYSCLK (1 and 3 switched in 3168) Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/pcm3168a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index ed0b4317c9ae..844602710eee 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -263,7 +263,7 @@ static unsigned int pcm3168a_scki_ratios[] = { #define PCM3168A_NUM_SCKI_RATIOS_DAC ARRAY_SIZE(pcm3168a_scki_ratios) #define PCM3168A_NUM_SCKI_RATIOS_ADC (ARRAY_SIZE(pcm3168a_scki_ratios) - 2) -#define PCM1368A_MAX_SYSCLK 36864000 +#define PCM3168A_MAX_SYSCLK 36864000 static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a) { @@ -296,7 +296,7 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; - if (freq > PCM1368A_MAX_SYSCLK) + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; ret = clk_set_rate(pcm3168a->scki, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links
by Stephan Gerhold 23 Jul '19

23 Jul '19

apq8016_sbc_parse_of() sets up multiple DAI links, depending on the number of nodes in the device tree. However, at the moment CPU and platform components are only allocated for the first link. This causes an oops when more than one link is defined: Internal error: Oops: 96000044 [#1] SMP CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4 Call trace: apq8016_sbc_platform_probe+0x1a8/0x3f0 platform_drv_probe+0x50/0xa0 ... Move the allocation inside the loop to ensure that each link is properly initialized. Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/apq8016_sbc.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index f60a71990f66..ac75838bbfab 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card) link = data->dai_link; - dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return ERR_PTR(-ENOMEM); + for_each_child_of_node(node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return ERR_PTR(-ENOMEM); - link->cpus = &dlc[0]; - link->platforms = &dlc[1]; + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; - link->num_cpus = 1; - link->num_platforms = 1; + link->num_cpus = 1; + link->num_platforms = 1; - for_each_child_of_node(node, np) { cpu = of_get_child_by_name(np, "cpu"); codec = of_get_child_by_name(np, "codec"); -- 2.22.0

3 2

[alsa-devel] [PATCH V4 0/2] recover the channel swap after xrun
by shengjiu.wang＠nxp.com 23 Jul '19

23 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> recover the channel swap after xrun Shengjiu Wang (2): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun sound/soc/fsl/fsl_esai.c | 266 ++++++++++++++++++++++++++++----------- 1 file changed, 193 insertions(+), 73 deletions(-) Changes in v4 -add more comments and remove read saisr Changes in V3 - update code sytle. Changes in v2 - add one patch for wrap operations to functions. - fix some coding style issue -- 2.21.0

3 5

[alsa-devel] [PATCH v2] ASoC: rt1308: Convert headers to SPDX
by shumingf＠realtek.com 23 Jul '19

23 Jul '19

From: Shuming Fan <shumingf(a)realtek.com> Convert include and source files to SPDX. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1308.c | 17 +++++++---------- sound/soc/codecs/rt1308.h | 6 ++---- 2 files changed, 9 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index cf3d72f99705..6f2ee6809dbb 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -1,13 +1,10 @@ -/* - * rt1308.c -- RT1308 ALSA SoC amplifier component driver - * - * Copyright 2019 Realtek Semiconductor Corp. - * Author: Derek Fang <derek.fang(a)realtek.com> - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// rt1308.c -- RT1308 ALSA SoC amplifier component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Derek Fang <derek.fang(a)realtek.com> +// #include <linux/module.h> #include <linux/moduleparam.h> diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h index c330aae1d527..ff7c423e879e 100644 --- a/sound/soc/codecs/rt1308.h +++ b/sound/soc/codecs/rt1308.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* - * RT1308.h -- RT1308 ALSA SoC amplifier component driver + * rt1308.h -- RT1308 ALSA SoC amplifier component driver * * Copyright 2019 Realtek Semiconductor Corp. * Author: Derek Fang <derek.fang(a)realtek.com> * - * 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 _RT1308_H_ -- 2.22.0

2 1

[alsa-devel] [PATCH] ASoC: rt5665: Fix a typo in the name of a function
by Christophe JAILLET 23 Jul '19

23 Jul '19

All function names start with rt5665_, except 'rt5655_set_verf()'. It is likely a typo. Fix it to be consistent. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/rt5665.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index f2ad3a4c3b7f..897e53d4c40a 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -2569,7 +2569,7 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static int rt5655_set_verf(struct snd_soc_dapm_widget *w, +static int rt5665_set_verf(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); @@ -2689,11 +2689,11 @@ static const struct snd_soc_dapm_widget rt5665_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5665_PWR_VOL, RT5665_PWR_MIC_DET_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5665_PWR_ANLG_1, RT5665_PWR_VREF1_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5665_PWR_ANLG_1, RT5665_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref3", RT5665_PWR_ANLG_1, RT5665_PWR_VREF3_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + rt5665_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5665_ASRC_1, -- 2.20.1

2 1

[alsa-devel] [PATCH v2 1/4] ALSA: compress: Fix regression on compressed capture streams
by Charles Keepax 23 Jul '19

23 Jul '19

A previous fix to the stop handling on compressed capture streams causes some knock on issues. The previous fix updated snd_compr_drain_notify to set the state back to PREPARED for capture streams. This causes some issues however as the handling for snd_compr_poll differs between the two states and some user-space applications were relying on the poll failing after the stream had been stopped. To correct this regression whilst still fixing the original problem the patch was addressing, update the capture handling to skip the PREPARED state rather than skipping the SETUP state as it has done until now. Fixes: 4f2ab5e1d13d ("ALSA: compress: Fix stop handling on compressed capture streams") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles include/sound/compress_driver.h | 5 +---- sound/core/compress_offload.c | 16 +++++++++++----- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index c5188ff724d12..bc88d6f964da9 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -173,10 +173,7 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) if (snd_BUG_ON(!stream)) return; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); } diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99b8821587053..d79aee6b9edd2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; -- 2.11.0

3 8

[alsa-devel] [PATCH] ASoC: dapm: fix a memory leak bug
by Wenwen Wang 23 Jul '19

23 Jul '19

From: Wenwen Wang <wenwen(a)cs.uga.edu> In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in dapm_cnew_widget() to hold the new dapm widget. Then, different actions are taken according to the id of the widget, i.e., 'w->id'. If any failure occurs during this process, snd_soc_dapm_new_control_unlocked() should be terminated by going to the 'request_failed' label. However, the allocated kernel buffer is not freed on this code path, leading to a memory leak bug. To fix the above issue, free the buffer before returning from snd_soc_dapm_new_control_unlocked() through the 'request_failed' label. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24..23b9b25 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3706,6 +3706,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); + kfree_const(w->sname); + kfree(w); return ERR_PTR(ret); } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/6] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 23 Jul '19

23 Jul '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://lwn.net/Articles/790651/ This patchset has been tested on SDM845 based DragonBoard DB845c with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Thanks, srini Srinivas Kandagatla (6): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings .../bindings/sound/qcom,wcd934x.txt | 188 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5979 +++++++++++++++++ sound/soc/codecs/wcd934x.h | 426 ++ 5 files changed, 6605 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.txt create mode 100644 sound/soc/codecs/wcd934x.c create mode 100644 sound/soc/codecs/wcd934x.h -- 2.21.0

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[alsa-devel] [PATCH] ASoC: intel: Add Broadwell rt5650 machine driver
by Jon Flatley 23 Jul '19

23 Jul '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 305 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 5 + 4 files changed, 323 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149818b5..c111ae177b4a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,17 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6445f90ea542..b5e2619607be 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_ obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 000000000000..cb875eeab055 --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,305 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(fe, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, + .init = bdw_rt5650_rtd_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(dummy, be, dummy) + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return snd_soc_register_card(&bdw_rt5650_card); +} + +static int bdw_rt5650_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bdw_rt5650_card); + return 0; +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .remove = bdw_rt5650_remove, + .driver = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index d27853e7a369..ba3d25658436 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -29,6 +29,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.22.0.657.g960e92d24f-goog

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[alsa-devel] [PATCH] ASoC: rt5677: remove superfluous set
by Curtis Malainey 22 Jul '19

22 Jul '19

We define this variable with the same value, no need to set it twice Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c779dc3474f9e..deaed5132dc95 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5098,7 +5098,6 @@ static irqreturn_t rt5677_irq(int unused, void *data) goto exit; } - irq_fired = false; for (i = 0; i < RT5677_IRQ_NUM; i++) { if (reg_irq & rt5677_irq_descs[i].status_mask) { irq_fired = true; -- 2.22.0.410.gd8fdbe21b5-goog

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