February 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Intel HDA / ALC662 analog surround problem
by Raymond Yau 06 Mar '15

06 Mar '15

> > >> > You have to post the output of alsa-info.sh > >> > >> Please see the attached file. > > > [ 0.655493] sound hdaudioC1D0: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:line > > [ 0.655494] sound hdaudioC1D0: speaker_outs=1 (0x15/0x0/0x0/0x0/0x0) > > [ 0.655495] sound hdaudioC1D0: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) > > [ 0.655496] sound hdaudioC1D0: mono: mono_out=0x0 > > [ 0.655496] sound hdaudioC1D0: dig-out=0x1e/0x0 > > [ 0.655497] sound hdaudioC1D0: inputs: > > [ 0.655498] sound hdaudioC1D0: Front Mic=0x19 > > [ 0.655499] sound hdaudioC1D0: Rear Mic=0x18 > > [ 0.655499] sound hdaudioC1D0: Line=0x1a > > > This is a known bug of hda_generic.c for those desktop with internal > > speaker and three audio jacks at rear panel (e.g. those lenovo thinkcenter > > with alc66x codec) > > > The driver prefer to assign volume control/dac to headphone, line out and > > internal speaker instead of line out and the other two multi io jacks > > > The workaround is to change the internal speaker as not connected node > > 0x15 > > Could you please advise how to do this? Is it possible from userland or do I > have to patch something? > > Would a new fixup mode definition for the kernel help and if yes, how should > it look like, i.e. which pins should be listed? (I'm capable of submitting a > kernel patch.) Try early patching/hdajackretask to change node 0x15 Speaker as not connected > > > or > > > You need to increase badness of multi io so that driver select multi io as > > the best config > > I've found your patch from 2013 while googling for the topic. The current > kernel already sets BAD_MULTI_IO = 0x120. > > Do you mean to further increase it? If so, to what value, or is increasing > possible from userland? > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… This patch has side effect on your configuration with three DAC as it create "Master playback volume" at node 0x02 and "Speaker+LO Playback switch at node 0x14 line out jack

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[alsa-devel] [PATCH] SoC: Fix MAX98357A codec driver dependencies
by Guenter Roeck 05 Mar '15

05 Mar '15

The max98357a driver depends on GPIOLIB. This may cause the following build failure. sound/soc/codecs/max98357a.c: In function 'max98357a_daiops_trigger': sound/soc/codecs/max98357a.c:30:3: error: implicit declaration of function 'gpiod_set_value' sound/soc/codecs/max98357a.c: In function 'max98357a_codec_probe': sound/soc/codecs/max98357a.c:55:2: error: implicit declaration of function 'devm_gpiod_get' sound/soc/codecs/max98357a.c:61:2: error: implicit declaration of function 'gpiod_direction_output' Seen with mips:allmodconfig as well as various randconfig builds. Fixes: af5adf129369 ("ASoC: max98357a: Add MAX98357A codec driver") Cc: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c1..ea9f0e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C - select SND_SOC_MAX98357A + select SND_SOC_MAX98357A if GPIOLIB select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C -- 2.1.0

2 3

[alsa-devel] [PATCH 1/3] ASoC: ak4671: Fix control-less DAPM routes
by Lars-Peter Clausen 05 Mar '15

05 Mar '15

Routes without a control must use NULL for the control name. The ak4671 driver uses "NULL" instead in a few places. Previous to commit 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") the DAPM core silently ignored non-NULL controls on non-mixer and non-mux routes. But starting with that commit it will complain and not add the route breaking the ak4671 driver in the process. This patch replaces the incorrect "NULL" control name with NULL to fix the issue. Fixes: 5fe5b767dc6f ("ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4671.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 632e89f..2a58b1d 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -343,25 +343,25 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { }; static const struct snd_soc_dapm_route ak4671_intercon[] = { - {"DAC Left", "NULL", "PMPLL"}, - {"DAC Right", "NULL", "PMPLL"}, - {"ADC Left", "NULL", "PMPLL"}, - {"ADC Right", "NULL", "PMPLL"}, + {"DAC Left", NULL, "PMPLL"}, + {"DAC Right", NULL, "PMPLL"}, + {"ADC Left", NULL, "PMPLL"}, + {"ADC Right", NULL, "PMPLL"}, /* Outputs */ - {"LOUT1", "NULL", "LOUT1 Mixer"}, - {"ROUT1", "NULL", "ROUT1 Mixer"}, - {"LOUT2", "NULL", "LOUT2 Mix Amp"}, - {"ROUT2", "NULL", "ROUT2 Mix Amp"}, - {"LOUT3", "NULL", "LOUT3 Mixer"}, - {"ROUT3", "NULL", "ROUT3 Mixer"}, + {"LOUT1", NULL, "LOUT1 Mixer"}, + {"ROUT1", NULL, "ROUT1 Mixer"}, + {"LOUT2", NULL, "LOUT2 Mix Amp"}, + {"ROUT2", NULL, "ROUT2 Mix Amp"}, + {"LOUT3", NULL, "LOUT3 Mixer"}, + {"ROUT3", NULL, "ROUT3 Mixer"}, {"LOUT1 Mixer", "DACL", "DAC Left"}, {"ROUT1 Mixer", "DACR", "DAC Right"}, {"LOUT2 Mixer", "DACHL", "DAC Left"}, {"ROUT2 Mixer", "DACHR", "DAC Right"}, - {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, - {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT2 Mix Amp", NULL, "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", NULL, "ROUT2 Mixer"}, {"LOUT3 Mixer", "DACSL", "DAC Left"}, {"ROUT3 Mixer", "DACSR", "DAC Right"}, @@ -381,18 +381,18 @@ static const struct snd_soc_dapm_route ak4671_intercon[] = { {"LIN2", NULL, "Mic Bias"}, {"RIN2", NULL, "Mic Bias"}, - {"ADC Left", "NULL", "LIN MUX"}, - {"ADC Right", "NULL", "RIN MUX"}, + {"ADC Left", NULL, "LIN MUX"}, + {"ADC Right", NULL, "RIN MUX"}, /* Analog Loops */ - {"LIN1 Mixing Circuit", "NULL", "LIN1"}, - {"RIN1 Mixing Circuit", "NULL", "RIN1"}, - {"LIN2 Mixing Circuit", "NULL", "LIN2"}, - {"RIN2 Mixing Circuit", "NULL", "RIN2"}, - {"LIN3 Mixing Circuit", "NULL", "LIN3"}, - {"RIN3 Mixing Circuit", "NULL", "RIN3"}, - {"LIN4 Mixing Circuit", "NULL", "LIN4"}, - {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + {"LIN1 Mixing Circuit", NULL, "LIN1"}, + {"RIN1 Mixing Circuit", NULL, "RIN1"}, + {"LIN2 Mixing Circuit", NULL, "LIN2"}, + {"RIN2 Mixing Circuit", NULL, "RIN2"}, + {"LIN3 Mixing Circuit", NULL, "LIN3"}, + {"RIN3 Mixing Circuit", NULL, "RIN3"}, + {"LIN4 Mixing Circuit", NULL, "LIN4"}, + {"RIN4 Mixing Circuit", NULL, "RIN4"}, {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, -- 1.8.0

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Re: [alsa-devel] [ASoC]fsl_ssi: fix of_property_read_u32_array return value check
by Mark Brown 04 Mar '15

04 Mar '15

On Sun, Feb 15, 2015 at 11:11:43PM +0100, Maciej S. Szmigiero wrote: > of_property_read_u32_array returns 0 on success, so the return value shouldn't be inverted twice, > first on assignment then in condition expression. This doesn't apply to current code, please check and resend. Please also keep your changelog under 80 columns as covered in SubmittingPatches.

2 2

[alsa-devel] [TINYCOMPRESS][PATCH] crec: support streaming output to stdout
by Richard Fitzgerald 04 Mar '15

04 Mar '15

From: Ammar Zahid Ali Syed <aali(a)opensource.wolfsonmicro.com> It's useful for testing to be able to stream the captured audio to stdout instead of having to go via a file. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Ammar Ali <aali(a)opensource.wolfsonmicro.com> --- crec.c | 73 ++++++++++++++++++++++++++++++++++++++++----------------------- 1 files changed, 46 insertions(+), 27 deletions(-) diff --git a/crec.c b/crec.c index e3f189f..258fade 100644 --- a/crec.c +++ b/crec.c @@ -77,6 +77,8 @@ static int verbose; static int file; +static FILE *finfo; +static bool streamed; static const unsigned int DEFAULT_CHANNELS = 1; static const unsigned int DEFAULT_RATE = 44100; @@ -151,7 +153,7 @@ static void size_wave_header(struct wave_header *header, uint32_t size) static void usage(void) { - fprintf(stderr, "usage: crec [OPTIONS] filename\n" + fprintf(stderr, "usage: crec [OPTIONS] [filename]\n" "-c\tcard number\n" "-d\tdevice node\n" "-b\tbuffer size\n" @@ -162,6 +164,8 @@ static void usage(void) "-C\tSpecify the number of channels (default %u)\n" "-R\tSpecify the sample rate (default %u)\n" "-F\tSpecify the format: S16_LE, S32_LE (default S16_LE)\n\n" + "If filename is not given the output is\n" + "written to stdout\n\n" "Example:\n" "\tcrec -c 1 -d 2 test.wav\n" "\tcrec -f 5 test.wav\n", @@ -180,7 +184,7 @@ static int print_time(struct compress *compress) fprintf(stderr, "ERR: %s\n", compress_get_error(compress)); return -1; } else { - printf("DSP recorded %jd.%jd\n", + fprintf(finfo, "DSP recorded %jd.%jd\n", (intmax_t)tstamp.tv_sec, (intmax_t)tstamp.tv_nsec*1000); } return 0; @@ -195,6 +199,10 @@ static int finish_record() if (!file) return -ENOENT; + /* can't rewind if streaming to stdout */ + if (streamed) + return 0; + /* Get amount of data written to file */ ret = lseek(file, 0, SEEK_END); if (ret < 0) @@ -256,22 +264,27 @@ void capture_samples(char *name, unsigned int card, unsigned int device, length = length * rate * (samplebits / 8) * channels; if (verbose) - printf("%s: entry, reading %u bytes\n", __func__, length); - - file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); - if (file == -1) { - fprintf(stderr, "Unable to open file '%s'\n", name); - exit(EXIT_FAILURE); - } + fprintf(finfo, "%s: entry, reading %u bytes\n", __func__, length); + if (!name) { + file = STDOUT_FILENO; + } else { + file = open(name, O_RDWR | O_CREAT, S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP); + if (file == -1) { + fprintf(stderr, "Unable to open file '%s'\n", name); + exit(EXIT_FAILURE); + } + } /* Write a header, will update with size once record is complete */ - init_wave_header(&header, channels, rate, samplebits); - written = write(file, &header, sizeof(header)); - if (written != sizeof(header)) { + if (!streamed) { + init_wave_header(&header, channels, rate, samplebits); + written = write(file, &header, sizeof(header)); + if (written != sizeof(header)) { fprintf(stderr, "Error writing output file header: %s\n", strerror(errno)); goto file_exit; - } + } + } memset(&codec, 0, sizeof(codec)); memset(&config, 0, sizeof(config)); @@ -299,7 +312,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, }; if (verbose) - printf("%s: Opened compress device\n", __func__); + fprintf(finfo, "%s: Opened compress device\n", __func__); size = config.fragment_size; buffer = malloc(size * config.fragments); @@ -308,15 +321,15 @@ void capture_samples(char *name, unsigned int card, unsigned int device, goto comp_exit; } - printf("Recording file %s On Card %u device %u, with buffer of %lu bytes\n", + fprintf(finfo, "Recording file %s On Card %u device %u, with buffer of %lu bytes\n", name, card, device, buffer_size); - printf("Format %u Channels %u, %u Hz\n", - codec.id, codec.ch_out, rate); + fprintf(finfo, "Codec %u Format %u Channels %u, %u Hz\n", + codec.id, codec.format, codec.ch_out, rate); compress_start(compress); if (verbose) - printf("%s: Capturing audio NOW!!!\n", __func__); + fprintf(finfo, "%s: Capturing audio NOW!!!\n", __func__); do { if (length && size > length - total_read) @@ -344,7 +357,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) { print_time(compress); - printf("%s: read %d\n", __func__, read); + fprintf(finfo, "%s: read %d\n", __func__, read); } } } while (!length || total_read < length); @@ -362,7 +375,7 @@ void capture_samples(char *name, unsigned int card, unsigned int device, } if (verbose) - printf("%s: exit success\n", __func__); + fprintf(finfo, "%s: exit success\n", __func__); free(buffer); close(file); @@ -379,7 +392,7 @@ file_exit: close(file); if (verbose) - printf("%s: exit failure\n", __func__); + fprintf(finfo, "%s: exit failure\n", __func__); exit(EXIT_FAILURE); } @@ -408,7 +421,7 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } - if (argc < 2) + if (argc < 1) usage(); verbose = 0; @@ -456,15 +469,21 @@ int main(int argc, char **argv) exit(EXIT_FAILURE); } } - if (optind >= argc) - usage(); - - file = argv[optind]; + if (optind >= argc) { + file = NULL; + finfo = fopen("/dev/null", "w"); + streamed = true; + } else { + file = argv[optind]; + finfo = stdout; + streamed = false; + } capture_samples(file, card, device, buffer_size, frag, length, rate, channels, format); - printf("Finish capturing... Close Normally\n"); + fprintf(finfo, "Finish capturing... Close Normally\n"); + exit(EXIT_SUCCESS); } -- 1.7.2.5

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[alsa-devel] [PATCH 0/4] Armada 38x and 370 audio
by Marcin Wojtas 04 Mar '15

04 Mar '15

Hi, I send a series of patches adding I2S - S/PDIF support for Marvell Armada 38x SoC. It is same controller as in Armada 370, however two quirks had to be taken into consideration: * new PLL frequency setting * exceptive pinout of I2S and S/PDIF formats Kirkwood-i2s driver was enhanced with obtaining mem resources by name. Sound interface was enabled for Armada 385 DB board, using simple-card DT binding for both I2S and S/PDIF. In addition to that minor fix was applied for Armada 370 audio DT description (stable-CC'ed). Any comments and remarks are welcome. Best regards, Marcin Marcin Wojtas (4): ASoC: kirkwood: enable Kirkwood driver for Armada 38x platforms ARM: mvebu: add audio I2S controller to Armada 38x Device Tree ARM: mvebu: add audio support to Armada 385 DB ARM: mvebu: fix routing for analog audio input of Armada 370 DB platform .../devicetree/bindings/sound/mvebu-audio.txt | 14 ++- arch/arm/boot/dts/armada-370-db.dts | 2 +- arch/arm/boot/dts/armada-388-db.dts | 69 +++++++++++ arch/arm/boot/dts/armada-38x.dtsi | 17 +++ sound/soc/kirkwood/kirkwood-i2s.c | 132 ++++++++++++++++++++- sound/soc/kirkwood/kirkwood.h | 2 + 6 files changed, 232 insertions(+), 4 deletions(-) -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ASoC: rt5670: Add disabled item in dmic pin enum
by Bard Liao 03 Mar '15

03 Mar '15

Currently, we will configure dmic related pin definition if pdata.dmic_en is true. However, there is no disable option in the enum. So, any dmic is used, all 3 dmic related pins will be configured. It may cause unexpected pin definition. This patch adds a disable item for each dmic enum and take it as default. So the driver will not set the pin configuration if we don't set dmicx_data_pin in platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.h | 3 +++ 1 file changed, 3 insertions(+) mode change 100755 => 100644 sound/soc/codecs/rt5670.h diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h old mode 100755 new mode 100644 index 21f8e18..0a67adb --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1950,17 +1950,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: sam9g20_wm8731: drop machine_is_xxx
by Alexandre Belloni 02 Mar '15

02 Mar '15

Atmel based boards can now only be used with device tree. Drop non DT initialization. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/sam9g20_wm8731.c | 68 ++++++++++++++++++---------------------- 1 file changed, 31 insertions(+), 37 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 531728975bbb..639a85d0999c 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,8 +46,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <asm/mach-types.h> - #include "../codecs/wm8731.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -197,9 +195,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) int ret; if (!np) { - if (!(machine_is_at91sam9g20ek() || - machine_is_at91sam9g20ek_2mmc())) - return -ENODEV; + return -ENODEV; } ret = atmel_ssc_set_audio(0); @@ -236,39 +232,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) card->dev = &pdev->dev; /* Parse device node info */ - if (np) { - ret = snd_soc_of_parse_card_name(card, "atmel,model"); - if (ret) - goto err; - - ret = snd_soc_of_parse_audio_routing(card, - "atmel,audio-routing"); - if (ret) - goto err; - - /* Parse codec info */ - at91sam9g20ek_dai.codec_name = NULL; - codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); - if (!codec_np) { - dev_err(&pdev->dev, "codec info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.codec_of_node = codec_np; - - /* Parse dai and platform info */ - at91sam9g20ek_dai.cpu_dai_name = NULL; - at91sam9g20ek_dai.platform_name = NULL; - cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); - if (!cpu_np) { - dev_err(&pdev->dev, "dai and pcm info missing\n"); - return -EINVAL; - } - at91sam9g20ek_dai.cpu_of_node = cpu_np; - at91sam9g20ek_dai.platform_of_node = cpu_np; - - of_node_put(codec_np); - of_node_put(cpu_np); + ret = snd_soc_of_parse_card_name(card, "atmel,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, + "atmel,audio-routing"); + if (ret) + goto err; + + /* Parse codec info */ + at91sam9g20ek_dai.codec_name = NULL; + codec_np = of_parse_phandle(np, "atmel,audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "codec info missing\n"); + return -EINVAL; + } + at91sam9g20ek_dai.codec_of_node = codec_np; + + /* Parse dai and platform info */ + at91sam9g20ek_dai.cpu_dai_name = NULL; + at91sam9g20ek_dai.platform_name = NULL; + cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "dai and pcm info missing\n"); + return -EINVAL; } + at91sam9g20ek_dai.cpu_of_node = cpu_np; + at91sam9g20ek_dai.platform_of_node = cpu_np; + + of_node_put(codec_np); + of_node_put(cpu_np); ret = snd_soc_register_card(card); if (ret) { -- 2.1.0

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[alsa-devel] Issue with snd-aloop and alsaloop in my embedded linux custom board
by Srinivasan S 02 Mar '15

02 Mar '15

Dear Lars, I hope you were providing solutions for alsa issues, I am running out of the below issue, hope could you please help me in resolving this issue or could you please redirect to the respective links where my queries can be posted & get the solutions for the problem The below is the virtual cards created after configuring snd-aloop in the linux kernel snd-aloop provided 2 pass-through devices as shown in the below card 0, device 0 card 0, device 1 But am unable to perform the loop back ie., what ever am playing am unable to record in the virtual device As per the logs below, am using the below commands to perform loopback, could you please let me know why am unable to perform the loopback with the below commands or please let me know am I missing any configurations aplay -D hw:0,0,0 play.wav arecord -D hw:0,1,0 record.wav root@am335x-evm:/# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 1: UDA1345TS [TI UDA1345TS], device 0: UDA134x uda134x-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 root@am335x-evm:/# Kindly do the needful as early as possible, Many Thanks in advance

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[alsa-devel] [PATCH] ASoC: wm8731: let codec to manage clock by itself
by Bo Shen 02 Mar '15

02 Mar '15

Let the wm8731 codec to manage clock by itself. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b115ed8..ecd8424 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -13,6 +13,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -389,6 +391,9 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -490,6 +495,9 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + if (clk_prepare_enable(wm8731->mclk)) + return -EINVAL; break; case SND_SOC_BIAS_PREPARE: break; @@ -508,6 +516,8 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + if (IS_ENABLED(CONFIG_COMMON_CLK)) + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -666,6 +676,15 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&spi->dev, "Failed to get MCLK\n"); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -717,6 +736,15 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + if (IS_ENABLED(CONFIG_COMMON_CLK)) { + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); -- 2.3.0.rc0

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