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[alsa-devel] Question about snd_pcm_sw_params_set_start_threshold
by Christian Gmeiner 03 Jul '10

03 Jul '10

Hi all, I am running into some troubles with my linux plugin for the vdr. I want to play audio in sync of video. My current solution is to buffer about 500ms of video on the output device (pts based FIFO). I can play audio with alsa, but sometimes I am not in sync with the video. So I want to set start threshold to 0 to disable automatic playback start. Instead I want to buffer audio and start playback manual, if pts of video is near current scr clock of output device. What are my options to manually start playback? thanks, -- Christian Gmeiner, MSc

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[alsa-devel] [PATCH] sis7019: fix capture issues with multiple periods per buffer
by David Dillow 03 Jul '10

03 Jul '10

When using a timing voice to clock out periods during capture, the driver would slowly loose synchronization and never catch up, eventually reaching a point where it no longer generated interrupts. To avoid this situation, the virtual period clocking was changed to shorten the next timing period when our timing voice falls too far behind the capture voice. In addition, the first virtual period for the timing voice was slightly too short, causing the timing voice to initially be ahead of the capture voice. While tracking down this problem, I noticed that the expected sample offset was being incorrectly initialized, causing an overrun to be incorrectly reported when the timing voice happened to be perfectly synchronized. Reported-by: Hans Schou <linux(a)schou.dk> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- This has survived overnight testing at 44.1 kHz/16 bit/mono without issue. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 9cc1b5a..614ff6e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -264,11 +264,13 @@ static void sis_update_voice(struct voice *voice) * if using small periods. * * If we're less than 9 samples behind, we're on target. + * Otherwise, shorten the next vperiod by the amount we've + * been delayed. */ if (sync > -9) voice->vperiod = voice->sync_period_size + 1; else - voice->vperiod = voice->sync_period_size - 4; + voice->vperiod = voice->sync_period_size + sync + 10; if (voice->vperiod < voice->buffer_size) { sis_update_sso(voice, voice->vperiod); @@ -736,7 +738,7 @@ static void sis_prepare_timing_voice(struct voice *voice, period_size = buffer_size; /* Initially, we want to interrupt just a bit behind the end of - * the period we're clocking out. 10 samples seems to give a good + * the period we're clocking out. 12 samples seems to give a good * delay. * * We want to spread our interrupts throughout the virtual period, @@ -747,7 +749,7 @@ static void sis_prepare_timing_voice(struct voice *voice, * * This is all moot if we don't need to use virtual periods. */ - vperiod = runtime->period_size + 10; + vperiod = runtime->period_size + 12; if (vperiod > period_size) { u16 tail = vperiod % period_size; u16 quarter_period = period_size / 4; @@ -776,7 +778,7 @@ static void sis_prepare_timing_voice(struct voice *voice, */ timing->flags |= VOICE_SYNC_TIMING; timing->sync_base = voice->ctrl_base; - timing->sync_cso = runtime->period_size - 1; + timing->sync_cso = runtime->period_size; timing->sync_period_size = runtime->period_size; timing->sync_buffer_size = runtime->buffer_size; timing->period_size = period_size;

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Re: [alsa-devel] [PATCH] - arecord - remove "front" and related devices from the output of list-pcm option of arecord
by Jaroslav Kysela 03 Jul '10

03 Jul '10

On Thu, 1 Jul 2010, Raymond Yau wrote: > Replace "front" device by "hw" device and remove "rear", "center_lfe" , > "side" , "surroundxx" from the output of "arecord -L" since "front" device > cannot be used for capturing and the front device is for playback only . NAK. Fix alsa-lib/configuration to return proper "IOID". Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH 3/3] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..5d8673b 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8987", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

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[alsa-devel] [PATCH 1/2] ASoC: Add SmartQ sound driver
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds sound support for the SmartQ board. The hardware consists of a S3C6410 coupled with a WM8987 over I²S. As there's no driver for the WM8987 currently, this uses the WM8750 driver (which should be register compatible). Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/soc/s3c24xx/Kconfig | 6 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smartq_wm8987.c | 300 +++++++++++++++++++++++++++++++++++++ 3 files changed, 308 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smartq_wm8987.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..4f46ba0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -125,3 +125,9 @@ config SND_SOC_SMDK_WM9713 select SND_S3C_SOC_AC97 help Sat Y if you want to add support for SoC audio on the SMDK. + +config SND_S3C64XX_SOC_SMARTQ + tristate "SoC I2S Audio support for SmartQ board" + depends on SND_S3C24XX_SOC && MACH_SMARTQ + select SND_S3C64XX_SOC_I2S + select SND_SOC_WM8750 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 81d8dc5..50172c3 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o +snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o +obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c new file mode 100644 index 0000000..ef6183f --- /dev/null +++ b/sound/soc/s3c24xx/smartq_wm8987.c @@ -0,0 +1,300 @@ +/* sound/soc/s3c24xx/smartq_wm8987.c + * + * Copyright 2010 Maurus Cuelenaere <mcuelenaere(a)gmail.com> + * + * Based on smdk6410_wm8987.c + * Copyright 2007 Wolfson Microelectronics PLC. - linux(a)wolfsonmicro.com + * Graeme Gregory - graeme.gregory(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#include "../codecs/wm8750.h" + +/* + * According to http://www.wolfsonmicro.com/products/WM8987 the WM8987 is + * register compatible with the WM8750, so I'm currently using that one as + * there's no WM8987 codec driver. + */ + +static struct snd_soc_card snd_soc_smartq; + +static int smartq_hifi_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->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret; + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + clk = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SmartQ WM8987 HiFi DAI operations. + */ +static struct snd_soc_ops smartq_hifi_ops = { + .hw_params = smartq_hifi_hw_params, +}; + +static struct snd_soc_jack smartq_jack; + +static struct snd_soc_jack_pin smartq_jack_pins[] = { + /* Disable speaker when headphone is plugged in */ + { + .pin = "Internal Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = true, + }, +}; + +static struct snd_soc_jack_gpio smartq_jack_gpios[] = { + { + .gpio = S3C64XX_GPL(12), + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; + +static const struct snd_kcontrol_new wm8987_smartq_controls[] = { + SOC_DAPM_PIN_SWITCH("Internal Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), +}; + +static int smartq_speaker_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LOUT2"}, + {"Headphone Jack", NULL, "ROUT2"}, + + {"Internal Speaker", NULL, "LOUT2"}, + {"Internal Speaker", NULL, "ROUT2"}, + + {"Mic Bias", NULL, "Internal Mic"}, + {"LINPUT2", NULL, "Mic Bias"}, +}; + +static int smartq_wm8987_init(struct snd_soc_codec *codec) +{ + int err = 0; + + /* Add SmartQ specific widgets */ + snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets, + ARRAY_SIZE(wm8987_dapm_widgets)); + + /* add SmartQ specific controls */ + err = snd_soc_add_controls(codec, wm8987_smartq_controls, + ARRAY_SIZE(wm8987_smartq_controls)); + + if (err < 0) + return err; + + /* setup SmartQ specific audio path */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "LINPUT1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "ROUT1"); + + /* set endpoints to default off mode */ + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Internal Mic"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; + + /* Headphone jack detection */ + err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), + smartq_jack_pins); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&smartq_jack, + ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + return err; +} + +static struct snd_soc_dai_link smartq_dai[] = { + { + .name = "wm8987", + .stream_name = "SmartQ Hi-Fi", + .cpu_dai = &s3c64xx_i2s_dai[0], + .codec_dai = &wm8750_dai, + .init = smartq_wm8987_init, + .ops = &smartq_hifi_ops, + }, +}; + +static struct snd_soc_card snd_soc_smartq = { + .name = "SmartQ", + .platform = &s3c24xx_soc_platform, + .dai_link = smartq_dai, + .num_links = ARRAY_SIZE(smartq_dai), +}; + +static struct snd_soc_device smartq_snd_devdata = { + .card = &snd_soc_smartq, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *smartq_snd_device; + +static int __init smartq_init(void) +{ + int ret; + + if (!machine_is_smartq7() && !machine_is_smartq5()) { + pr_info("Only SmartQ is supported by this ASoC driver\n"); + return -ENODEV; + } + + smartq_snd_device = platform_device_alloc("soc-audio", -1); + if (!smartq_snd_device) + return -ENOMEM; + + platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata); + smartq_snd_devdata.dev = &smartq_snd_device->dev; + + ret = platform_device_add(smartq_snd_device); + if (ret) { + platform_device_put(smartq_snd_device); + return ret; + } + + /* Initialise GPIOs used by amplifiers */ + ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown"); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n"); + goto err_unregister_device; + } + + /* Disable amplifiers */ + ret = gpio_direction_output(S3C64XX_GPK(12), 1); + if (ret) { + dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n"); + goto err_free_gpio_amp_shut; + } + + dev_info(&smartq_snd_device->dev, "SmartQ WM8987 audio driver inited\n"); + + return 0; + +err_free_gpio_amp_shut: + gpio_free(S3C64XX_GPK(12)); +err_unregister_device: + platform_device_unregister(smartq_snd_device); + + return ret; +} + +static void __exit smartq_exit(void) +{ + snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios), + smartq_jack_gpios); + + platform_device_unregister(smartq_snd_device); +} + +module_init(smartq_init); +module_exit(smartq_exit); + +/* Module information */ +MODULE_AUTHOR("Maurus Cuelenaere"); +MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

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[alsa-devel] [PATCH 2/2] ARM: S3C64XX: Add audio support to SmartQ
by Maurus Cuelenaere 03 Jul '10

03 Jul '10

This adds the I²C board information for the WM8987 used in the SmartQ as audio codec and adds the I²C/I²S platform drivers. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- arch/arm/mach-s3c64xx/mach-smartq.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-s3c64xx/mach-smartq.c b/arch/arm/mach-s3c64xx/mach-smartq.c index 028d080..ee77255 100644 --- a/arch/arm/mach-s3c64xx/mach-smartq.c +++ b/arch/arm/mach-s3c64xx/mach-smartq.c @@ -199,6 +199,9 @@ static struct platform_device smartq_lcd_power_device = { .dev.platform_data = &smartq_lcd_power_data, }; +static struct i2c_board_info smartq_i2c_devs[] __initdata = { + { I2C_BOARD_INFO("wm8750", 0x1a), }, +}; static struct platform_device *smartq_devices[] __initdata = { &s3c_device_hsmmc1, /* Init iNAND first, ... */ @@ -213,6 +216,7 @@ static struct platform_device *smartq_devices[] __initdata = { &s3c_device_timer[1], &s3c_device_ts, &s3c_device_usb_hsotg, + &s3c64xx_device_iis0, &smartq_backlight_device, &smartq_lcd_power_device, &smartq_usb_otg_vbus_dev, @@ -354,6 +358,9 @@ void __init smartq_machine_init(void) s3c_sdhci2_set_platdata(&smartq_internal_hsmmc_pdata); s3c24xx_ts_set_platdata(&smartq_touchscreen_pdata); + i2c_register_board_info(0, smartq_i2c_devs, + ARRAY_SIZE(smartq_i2c_devs)); + WARN_ON(smartq_power_off_init()); WARN_ON(smartq_usb_host_init()); WARN_ON(smartq_usb_otg_init()); -- 1.7.0.4

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[alsa-devel] alsamixer-qt4 version 0.3 released
by Sebastian H. 01 Jul '10

01 Jul '10

Hello again The new version 0.3.0 of alsamixer-qt4 is available. Check it out at http://xwmw.org/alsamixer-qt4/ The files are at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ Regards, Sebastian H.

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[alsa-devel] [PATCH v3] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 01 Jul '10

01 Jul '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 o remove "default volume" o remove explain from commit log sound/soc/codecs/ak4642.c | 24 +++++++++++++++++++++--- 1 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..b41599d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -548,6 +563,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

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[alsa-devel] wrong decibel data?
by Nicolo' Chieffo 01 Jul '10

01 Jul '10

Hello, I have a laptop with an intel hda audio chipset (STAC92xx) My problem is that when I put pulseaudio volume to 14% (or less) I can't hear any audio coming from speakers or headphones: in fact the alsamixer volume is set to 0% I already filed a bug to pulseaudio [1] in which the developers say that this is a common problem which is usually caused by the ALSA audio driver which reports a wrong decibel value. [1] http://pulseaudio.org/ticket/580

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[alsa-devel] [PATCH 3/3] ASoC: DaVinci: Added fast clock timing for McBSP (I2S)
by Raffaele Recalcati 01 Jul '10

01 Jul '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> i2s_fast_clock switch can be used to have a better approximate frequency. The waveform will be not symmetric. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++++ sound/soc/davinci/davinci-i2s.c | 24 +++++++++++++++++++----- 2 files changed, 26 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 314570d..2d8f8af 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not symmetrical, but sampling + * frequency is better approximated + */ + bool i2s_fast_clock; + /* To be used when cpu gets clock from external pin */ int clk_input_pin; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e478be9..0af5bae 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -150,6 +150,7 @@ struct davinci_mcbsp_dev { */ unsigned enable_channel_combine:1; + bool i2s_fast_clock; unsigned int fmt; int clk_div; int clk_input_pin; @@ -443,11 +444,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SRGR_CLKSM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); - /* symmetric waveforms */ - clk_div = freq / (mcbsp_word_length * 16) / - params->rate_num * params->rate_den; - srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * - 16 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } clk_div &= 0xFF; srgr |= clk_div; break; @@ -643,6 +656,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

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