[alsa-devel] [PATCH] ARM: pxa: add support for Raumfeld DDX
This new product features a STA32x codec. To support it, some rework of the regulator initialization for the Raumfeld platform is necessary.
Signed-off-by: Daniel Mack zonque@gmail.com Cc: Eric Miao eric.y.miao@gmail.com Cc: Mark Brown broonie@opensource.wolfsonmicro.com Cc: Liam Girdwood lrg@ti.com ---
Mark, Liam, Eric - which tree would be good to the this merged?
arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 249 +++++++++++++++++++++++++++++++++++++++--- 3 files changed, 338 insertions(+), 32 deletions(-)
diff --git a/arch/arm/mach-pxa/raumfeld.c b/arch/arm/mach-pxa/raumfeld.c index 7856fe4..22366ff 100644 --- a/arch/arm/mach-pxa/raumfeld.c +++ b/arch/arm/mach-pxa/raumfeld.c @@ -850,7 +850,7 @@ struct regulator_init_data audio_va_initdata = { }, };
-static struct fixed_voltage_config audio_va_config = { +static struct fixed_voltage_config cs4270_va_config = { .supply_name = "audio_va", .microvolts = 5000000, .gpio = GPIO_AUDIO_VA_ENABLE, @@ -859,47 +859,114 @@ static struct fixed_voltage_config audio_va_config = { .init_data = &audio_va_initdata, };
-static struct platform_device audio_va_device = { +static struct platform_device cs4270_va_device = { .name = "reg-fixed-voltage", .id = 0, .dev = { - .platform_data = &audio_va_config, + .platform_data = &cs4270_va_config, }, };
/* Dummy supplies for Codec's VD/VLC */
-static struct regulator_consumer_supply audio_dummy_supplies[] = { +static struct regulator_consumer_supply cs4270_dummy_supplies[] = { REGULATOR_SUPPLY("vd", "0-0048"), REGULATOR_SUPPLY("vlc", "0-0048"), };
-struct regulator_init_data audio_dummy_initdata = { - .consumer_supplies = audio_dummy_supplies, - .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), +struct regulator_init_data cs4270_dummy_initdata = { + .consumer_supplies = cs4270_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(cs4270_dummy_supplies), .constraints = { .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, };
-static struct fixed_voltage_config audio_dummy_config = { - .supply_name = "audio_vd", +static struct fixed_voltage_config cs4270_dummy_config = { + .supply_name = "cs4270_vd", .microvolts = 3300000, .gpio = -1, - .init_data = &audio_dummy_initdata, + .init_data = &cs4270_dummy_initdata, };
-static struct platform_device audio_supply_dummy_device = { +static struct platform_device cs4270_supply_dummy_device = { .name = "reg-fixed-voltage", .id = 1, .dev = { - .platform_data = &audio_dummy_config, + .platform_data = &cs4270_dummy_config, }, };
-static struct platform_device *audio_regulator_devices[] = { - &audio_va_device, - &audio_supply_dummy_device, +static struct platform_device *cs4270_regulator_devices[] = { + &cs4270_va_device, + &cs4270_supply_dummy_device, +}; + +/* Fixed regulator for sta32x Vdda supply + * 0-001a maps to the sta32x codec devname (derived from i2c bus num + addr) + */ + +static struct regulator_consumer_supply sta32x_va_consumer_supply = + REGULATOR_SUPPLY("Vdda", "0-001a"); + +struct regulator_init_data sta32x_va_initdata = { + .consumer_supplies = &sta32x_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_va_config = { + .supply_name = "audio_va", + .microvolts = 3300000, + .gpio = GPIO_AUDIO_VA_ENABLE, + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &sta32x_va_initdata, +}; + +static struct platform_device sta32x_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &sta32x_va_config, + }, +}; + +/* Dummy supplies for sta32x codec's Vdd3/Vcc */ + +static struct regulator_consumer_supply sta32x_dummy_supplies[] = { + REGULATOR_SUPPLY("Vdd3", "0-001a"), + REGULATOR_SUPPLY("Vcc", "0-001a"), +}; + +struct regulator_init_data sta32x_dummy_initdata = { + .consumer_supplies = sta32x_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(sta32x_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &sta32x_dummy_initdata, +}; + +static struct platform_device sta32x_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &sta32x_dummy_config, + }, +}; + +static struct platform_device *sta32x_regulator_devices[] = { + &sta32x_va_device, + &sta32x_supply_dummy_device, };
/** @@ -948,6 +1015,11 @@ static struct i2c_board_info raumfeld_connector_i2c_board_info __initdata = { .addr = 0x48, };
+static struct i2c_board_info raumfeld_ddx_i2c_board_info __initdata = { + .type = "sta326", + .addr = 0x1a, +}; + static struct eeti_ts_platform_data eeti_ts_pdata = { .irq_active_high = 1, }; @@ -987,7 +1059,10 @@ static void __init raumfeld_audio_init(void) else gpio_direction_output(GPIO_MCLK_RESET, 1);
- platform_add_devices(ARRAY_AND_SIZE(audio_regulator_devices)); + if ((system_rev & 0xff00) == 0x0400) + platform_add_devices(ARRAY_AND_SIZE(sta32x_regulator_devices)); + else + platform_add_devices(ARRAY_AND_SIZE(cs4270_regulator_devices)); }
static void __init raumfeld_common_init(void) @@ -1060,7 +1135,11 @@ static void __init raumfeld_connector_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_connector_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(connector_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1);
platform_device_register(&smc91x_device);
@@ -1072,7 +1151,11 @@ static void __init raumfeld_speaker_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_speaker_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(speaker_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1);
platform_device_register(&smc91x_device); platform_device_register(&rotary_encoder_device); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 + select SND_SOC_STA32X + select SND_SOC_WM8782 help Say Y if you want to add support for SoC audio on Raumfeld devices
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1..894731b 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/soc.h>
#include <asm/mach-types.h> @@ -225,6 +226,177 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, };
+/* STA32X */ +static int raumfeld_sta32x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* FIXME: currently only fixed MCLK of 12.288MHz is available */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000); + + /* PXA DMA cannot do zero extend for 24bit samples, + * thus only 16bit (two samples packet into 32bit word) + * or 32bit samples are possible + */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE); + + /* FIXME: we have a fixed MCLK, set it here so ALSA knows + * the supported sample rates and can resample if necessary + */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 12288000, 0); +} + +static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} + +static int raumfeld_sta32x_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; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 32000: + case 48000: + case 96000: + case 192000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + /* this enables network mode for 2 * 32bit samples */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 0, 2, 32); + if (ret < 0) + return ret; + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + /* this disables network mode */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0, 0, 16); + if (ret < 0) + return ret; + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_sta32x_ops = { + .startup = raumfeld_sta32x_startup, + .shutdown = raumfeld_sta32x_shutdown, + .hw_params = raumfeld_sta32x_hw_params, +}; + +/* WM8782 */ +static int raumfeld_wm8782_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 *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // FIXME: currently only fixed MCLK of 12.288MHz is available + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_wm8782_ops = { + .hw_params = raumfeld_wm8782_hw_params, +}; + #define DAI_LINK_CS4270 \ { \ .name = "CS4270", \ @@ -247,6 +419,28 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .codec_name = "spi0.0", \ }
+#define DAI_LINK_STA32X \ +{ \ + .name = "STA32X", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "STA32X", \ + .ops = &raumfeld_sta32x_ops, \ + .codec_name = "sta32x.0-001a", \ +} + +#define DAI_LINK_WM8782 \ +{ \ + .name = "wm8782", \ + .stream_name = "wm8782", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "wm8782", \ + .codec_name = "wm8782.0", \ + .ops = &raumfeld_wm8782_ops, \ +} + static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, @@ -258,6 +452,12 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = DAI_LINK_CS4270, };
+static struct snd_soc_dai_link snd_soc_raumfeld_ddx_dai[] = +{ + DAI_LINK_STA32X, + DAI_LINK_WM8782, +}; + static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", .dai_link = snd_soc_raumfeld_connector_dai, @@ -274,7 +474,15 @@ static struct snd_soc_card snd_soc_raumfeld_speaker = { .resume_pre = raumfeld_analog_resume, };
-static struct platform_device *raumfeld_audio_device; +static struct snd_soc_card snd_soc_raumfeld_ddx = { + .name = "Raumfeld DDX", + .dai_link = snd_soc_raumfeld_ddx_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_ddx_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct platform_device *raumfeld_audio_device, *wm8782_device;
static int __init raumfeld_audio_init(void) { @@ -284,26 +492,39 @@ static int __init raumfeld_audio_init(void) !machine_is_raumfeld_connector()) return 0;
- max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); + if ((system_rev & 0xff00) == 0x0400) { + wm8782_device = platform_device_alloc("wm8782", 0); + if (!wm8782_device) + return -ENOMEM;
- if (!max9486_client) - return -ENOMEM; + platform_device_add(wm8782_device); + } else { + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM;
- set_max9485_clk(MAX9485_MCLK_FREQ_122880); + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + }
- /* Register analog device */ + /* Register audio device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM;
- if (machine_is_raumfeld_speaker()) + if ((system_rev & 0xff00) == 0x0400) platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); + &snd_soc_raumfeld_ddx); + else { + if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + }
ret = platform_device_add(raumfeld_audio_device); if (ret < 0) @@ -317,8 +538,8 @@ static void __exit raumfeld_audio_exit(void) { raumfeld_enable_audio(false);
+ platform_device_unregister(wm8782_device); platform_device_unregister(raumfeld_audio_device); - i2c_unregister_device(max9486_client);
gpio_free(GPIO_MCLK_RESET);
On Mon, Nov 07, 2011 at 11:26:02AM +0100, Daniel Mack wrote:
arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 249 +++++++++++++++++++++++++++++++++++++++---
Is there any cross dependency between these bits? I honestly didn't notice the ASoC bit in here...
index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104
- select SND_SOC_STA32X
- select SND_SOC_WM8782
Is this really a single driver? Looking at the code it looks like there's little if any code sharing between the different machines, it'd help with maintainability to have multiple simple drivers.
- /* PXA DMA cannot do zero extend for 24bit samples,
- thus only 16bit (two samples packet into 32bit word)
- or 32bit samples are possible
- */
- snd_pcm_hw_constraint_mask64(substream->runtime,
SNDRV_PCM_HW_PARAM_FORMAT,SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE);
If the PXA DMA controller can't do 24 bit audio it should be imposing this constraint.
+static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- /* set freq to 0 to enable all possible codec sample rates */
- snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
Are you sure this does the right thing with simultaneous playback and record?
- fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |SND_SOC_DAIFMT_CBS_CFS;
Set this using dai_fmt in hte machine driver.
On 11/15/2011 03:13 PM, Mark Brown wrote:
On Mon, Nov 07, 2011 at 11:26:02AM +0100, Daniel Mack wrote:
arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 249 +++++++++++++++++++++++++++++++++++++++---
Is there any cross dependency between these bits?
You're right - I assumed there are, but it seems that I was mistaken. I'll split this up and re-submit, so it can go trhough appropriate channels.
I honestly didn't notice the ASoC bit in here...
That's at least parially my fault, as something in my work flow crippled the email addresses in the patch, so git send-email didn't actually put you on Cc:.
index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104
- select SND_SOC_STA32X
- select SND_SOC_WM8782
Is this really a single driver? Looking at the code it looks like there's little if any code sharing between the different machines, it'd help with maintainability to have multiple simple drivers.
Hmm. This new version of Raumfeld's hardware features a new ADC and DAC, and the configuration to use is determined by looking at the system revision. If this would be split into multiple smaller drivers, there would be need for a 'master' to dispatch the possible options. I think I'll do that once the next generation is about to land, and leave it as it is for now. Ok for you?
- /* PXA DMA cannot do zero extend for 24bit samples,
- thus only 16bit (two samples packet into 32bit word)
- or 32bit samples are possible
- */
- snd_pcm_hw_constraint_mask64(substream->runtime,
SNDRV_PCM_HW_PARAM_FORMAT,SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE);If the PXA DMA controller can't do 24 bit audio it should be imposing this constraint.
Right. Will fix this in the CPU DAI.
+static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- /* set freq to 0 to enable all possible codec sample rates */
- snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
+}
Are you sure this does the right thing with simultaneous playback and record?
Yes, this works fine. We use this code since awhile already.
- fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |SND_SOC_DAIFMT_CBS_CFS;Set this using dai_fmt in hte machine driver.
Can you elaborate on this one? I couldn't find an example for this, sorry.
Daniel
On Wed, Nov 16, 2011 at 01:56:47PM +0100, Daniel Mack wrote:
On 11/15/2011 03:13 PM, Mark Brown wrote:
Is this really a single driver? Looking at the code it looks like there's little if any code sharing between the different machines, it'd help with maintainability to have multiple simple drivers.
Hmm. This new version of Raumfeld's hardware features a new ADC and DAC, and the configuration to use is determined by looking at the system revision. If this would be split into multiple smaller drivers, there would be need for a 'master' to dispatch the possible options. I think
No, not really.
I'll do that once the next generation is about to land, and leave it as it is for now. Ok for you?
What I'd expect here is that the arch/arm code would register a different platform device depending on which board it's running on (though actually controlling I2C device registration is enough as ASoC won't probe the sound driver until the components appear).
- fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |SND_SOC_DAIFMT_CBS_CFS;
Set this using dai_fmt in hte machine driver.
Can you elaborate on this one? I couldn't find an example for this, sorry.
Look at the snd_soc_dai_link definition, and things like speyside.
On 11/16/2011 02:19 PM, Mark Brown wrote:
On Wed, Nov 16, 2011 at 01:56:47PM +0100, Daniel Mack wrote:
On 11/15/2011 03:13 PM, Mark Brown wrote:
Is this really a single driver? Looking at the code it looks like there's little if any code sharing between the different machines, it'd help with maintainability to have multiple simple drivers.
Hmm. This new version of Raumfeld's hardware features a new ADC and DAC, and the configuration to use is determined by looking at the system revision. If this would be split into multiple smaller drivers, there would be need for a 'master' to dispatch the possible options. I think
No, not really.
I'll do that once the next generation is about to land, and leave it as it is for now. Ok for you?
What I'd expect here is that the arch/arm code would register a different platform device depending on which board it's running on (though actually controlling I2C device registration is enough as ASoC won't probe the sound driver until the components appear).
Ok, I can do this. However, the two driver will share some logic wrt GPIO lines that put the codecs into reset. But ok, we can live with that I guess, as it in fact makes the driver smaller and easier to read.
- fmt = SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |SND_SOC_DAIFMT_CBS_CFS;Set this using dai_fmt in hte machine driver.
Can you elaborate on this one? I couldn't find an example for this, sorry.
Look at the snd_soc_dai_link definition, and things like speyside.
Sorry, I still don't get it. All implementations I can find call snd_soc_dai_set_fmt() for the cpu and codec dais from the .ops->hw_params function they reference from their snd_soc_dai_links. Just like this new code does as well. What am I missing?
Thanks for your review,
Daniel
On Wed, Nov 16, 2011 at 03:06:46PM +0100, Daniel Mack wrote:
On 11/16/2011 02:19 PM, Mark Brown wrote:
What I'd expect here is that the arch/arm code would register a different platform device depending on which board it's running on (though actually controlling I2C device registration is enough as ASoC won't probe the sound driver until the components appear).
Ok, I can do this. However, the two driver will share some logic wrt GPIO lines that put the codecs into reset. But ok, we can live with that I guess, as it in fact makes the driver smaller and easier to read.
Hrm, I'd expect that that logic would be in the CODEC drivers - they really do need to know that their registers went away.
Look at the snd_soc_dai_link definition, and things like speyside.
Sorry, I still don't get it. All implementations I can find call snd_soc_dai_set_fmt() for the cpu and codec dais from the .ops->hw_params function they reference from their snd_soc_dai_links. Just like this new code does as well. What am I missing?
Are you *sure* you're looking at current code? The dai_fmt field isn't that obscurely named...
On 11/16/2011 03:11 PM, Mark Brown wrote:
On Wed, Nov 16, 2011 at 03:06:46PM +0100, Daniel Mack wrote:
Sorry, I still don't get it. All implementations I can find call snd_soc_dai_set_fmt() for the cpu and codec dais from the .ops->hw_params function they reference from their snd_soc_dai_links. Just like this new code does as well. What am I missing?
Are you *sure* you're looking at current code? The dai_fmt field isn't that obscurely named...
Grrr. Sorry for that. We're based on 3.1 here, I should have noticed. I'll rebase, do all necessary cleanups and resubmit.
Daniel
On Wed, Nov 16, 2011 at 01:56:47PM +0100, Daniel Mack wrote:
On 11/15/2011 03:13 PM, Mark Brown wrote:
On Mon, Nov 07, 2011 at 11:26:02AM +0100, Daniel Mack wrote:
- /* PXA DMA cannot do zero extend for 24bit samples,
- thus only 16bit (two samples packet into 32bit word)
- or 32bit samples are possible
- */
- snd_pcm_hw_constraint_mask64(substream->runtime,
SNDRV_PCM_HW_PARAM_FORMAT,SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE);If the PXA DMA controller can't do 24 bit audio it should be imposing this constraint.
Right. Will fix this in the CPU DAI.
IIRC the PXA can handle 24bit audio, and the STA32x can handle it, too. However due to constraints in the clock ratios on the Raumfeld board this mode cannot be used. Thus this constraint is in sound/soc/pxa/raumfeld.c.
Does that make sense? It's been a while since I worked on this code, maybe I misremember. And I failed to mention it in the code comment :-(
Johannes
On Wed, Nov 16, 2011 at 04:07:22PM +0100, Johannes Stezenbach wrote:
IIRC the PXA can handle 24bit audio, and the STA32x can handle it, too. However due to constraints in the clock ratios on the Raumfeld board this mode cannot be used. Thus this constraint is in sound/soc/pxa/raumfeld.c.
If that's the case then the comment needs to be updated as currently what it says is that this is a CPU side limitation.
participants (3)
-
Daniel Mack -
Johannes Stezenbach -
Mark Brown