January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: soc_jack - add function to determine jack type
by Harsha Priya 01 Feb '11

01 Feb '11

This patch adds a helper function that determines the jack type given the mic bias value. For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 15 +++++++++++++++ sound/soc/soc-jack.c | 18 ++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaef..afe98eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -246,6 +246,19 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum soc_mic_bias_zones { + /* mic bias volutage range for Headphones*/ + SND_SOC_JACK_HP_START = 0, + SND_SOC_JACK_HP_END = 400, + /* mic bias volutage range for American Headset*/ + SND_SOC_JACK_AM_HS_START = 400, + SND_SOC_JACK_AM_HS_END = 650, + /* mic bias volutage range for Headset*/ + SND_SOC_JACK_HS_START = 650, + SND_SOC_JACK_HS_END = 2000, + SND_SOC_UNDEFINED, +}; + enum snd_soc_control_type { SND_SOC_CUSTOM, SND_SOC_I2C, @@ -307,6 +320,8 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_get_type(int mic_bias); + #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..462965a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -186,6 +186,24 @@ void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack + * + * @mic_bias: mic bias value detected + * + * Based on the mic bias value passed, this function helps identify + * the type of jack as either SND_JACK_HEADSET or SND_JACK_HEADPHONE. + */ +int snd_soc_jack_get_type(int mic_bias) +{ + if (mic_bias >= SND_SOC_JACK_HP_START && mic_bias < SND_SOC_JACK_HS_END) + return SND_JACK_HEADSET; + else + return SND_JACK_HEADPHONE; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + #ifdef CONFIG_GPIOLIB /* gpio detect */ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: Support 8 bit samples in soc-util
by Mark Brown 01 Feb '11

01 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-utils.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 1d07b93..e31d843 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -29,6 +29,10 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) int sample_size; switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_U8: + sample_size = 8; + break; case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: sample_size = 16; -- 1.7.2.3

2 2

[alsa-devel] Questions about Multi-streaming playback with HDA codec ad1988b
by Raymond Yau 01 Feb '11

01 Feb '11

The patch is to provide a new model which front panel headphone of a 6stack-dig desktop can be used for playback a stereo stream independent of the rear panel ad1988b is a HDA codec support 10 channel ( enough DAC for stereo + surround71 ) 1) Is this approach to create a new model correct or just need to implement a switch for activatie the "independent head phone" like those VIA codecs ? struct ad198x_spec { + hda_nid_t *alt_dac_nid; + struct hda_pcm_stream *stream_analog_alt_playback; } +static hda_nid_t ad1988_6stack_alt_dac_nid[1] = { + 0x03 +}; static const char *ad1988_models[AD1988_MODEL_LAST] = { [AD1988_6STACK] = "6stack", [AD1988_6STACK_DIG] = "6stack-dig", + [AD1988_6STACK_DIG_FP] = "6stack-dig-fp", [AD1988_3STACK] = "3stack", [AD1988_3STACK_DIG] = "3stack-dig", [AD1988_LAPTOP] = "laptop", [AD1988_LAPTOP_DIG] = "laptop-dig", [AD1988_AUTO] = "auto", }; enum { AD1988_6STACK, AD1988_6STACK_DIG, + AD1988_6STACK_DIG_FP, AD1988_3STACK, AD1988_3STACK_DIG, AD1988_LAPTOP, AD1988_LAPTOP_DIG, AD1988_AUTO, AD1988_MODEL_LAST, }; 2) do the driver need to define .prepare and .cleanup function or just need to use "hda_pcm_default_prepare() and hda_pcm_default_cleanup() static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, 0, format); return 0; } static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) { struct ad198x_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]); return 0; } static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in ad198x_build_pcms */ .ops = { .prepare = ad198x_alt_playback_pcm_prepare, .cleanup = ad198x_alt_playback_pcm_cleanup }, }; 3) Is it correct to use device 2 for analog alternate playback since some HDA codec implement this feature on "hw:0,0,1" and realtek driver create alternate capture on "hw:0,2" , this seem create a lot of problem of the application (e.g. PA server ) ? static int ad198x_build_pcms(struct hda_codec *codec) { if (spec->alt_dac_nid && spec->stream_analog_alt_playback) { codec->num_pcms++; info = spec->pcm_rec + 2; info->name = "AD198x Alternate Analog"; info->pcm_type = HDA_PCM_TYPE_AUDIO; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *spec->stream_analog_alt_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->alt_dac_nid[0]; } } static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), { } /* end */ }; static struct hda_verb ad1988_6stack_fp_init_verbs[] = { /* Front, Surround, CLFE, side DAC; unmute as default */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Headphone; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC:03h */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Port-D line-out path */ {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-F surround path */ {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-G CLFE path */ {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-H side path */ {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Mono out path */ {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ /* Port-B front mic-in path */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Port-C line-in path */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Port-E mic-in path */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Analog CD Input */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; static int patch_ad1988(struct hda_codec *codec) { switch (board_config) { case AD1988_6STACK_DIG_FP: spec->multiout.max_channels = 8; spec->multiout.num_dacs = 4; spec->multiout.dac_nids = ad1988_6stack_dac_nids; spec->alt_dac_nid = ad1988_6stack_alt_dac_nid; spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback; spec->input_mux = &ad1988_6stack_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1988_6stack_fp_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1988_6stack_fp_init_verbs; spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; spec->dig_in_nid = AD1988_SPDIF_IN; break; 3) What functions are still missing in this model ? a), do this new model need to disable hp_automute() function b) any change in ad198x_power_eapd_xxx() functions ? c) idenifty ad1988a and ad1988b

2 2

[alsa-devel] Problem with VIA VT1708S and git versions of alsa-driver.
by Mark Goldstein 31 Jan '11

31 Jan '11

Hello, Recently I'm having some strange issues with my VIA VT1708S on-board sound card. The description might be a bit long, so I'll give a brief summary first. I have ASUS P5QL/EPU motherboard with VT1708S 8-channel HDA Codec. There are 3 OSs installed on this machine: openSUSE 11.1 (my main system), openSUSE 11.3, Kubuntu 10.04. Audio worked fine until some updates at the beginning of September. Since then I have 2 different issues. 1).On 11.1 - Line In control seems inactive, instead Front Mic control controls the level from Line In jack. Front Mic is constanly disabled. alsaconf does not show hda-intel in the list of detected cards. Latest version it happened with is git20100925. When I compiled stable 1.0.23 sources, everything started working as expected again. I reported this issue on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 2) On oS 11.3 I start experiencing problems recording audio either from Mic or from Line. After some experiments I found out that the Capture1 control is somehow broken. Mute switch is reversed. When it is unmuted, there is no audio recorded. When it is muted, recording works. The same happened to at least one more user of openSUSE, see the thread "How to investigate the problem with MIC" on alsa-user mailing list. I've also added my comments to issue https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 but now I think it is different issue. Even when I have audio recorded normally when Capture1 is muted, I still have very distorted voice from Mic in Skype. It worked fine before the problem started. More details: On oS 11.1 I had my audio working fine after updating to 1.0.23 versions of alsa driver. I'm using opensuse/multimedia: repository that provides binary rpm of alsa-driver git snapshots. I have only one current kernel 2.6.27.48 on 11.1. On oS 11.3 audio worked right out of the box, since it uses kernels with alsa-driver 1.0.23. I have 2-3 kernels installed at the same time - normal 2.6.34 from official repository and 2.6.35 from kernel build service. I'm not sure when exactly the problems started, but definitely at the beginning of September. I've initially saw that on oS 11.1 I can't hear the audio from my TV card connected to Line In. Then I noticed that Mic does not work in Skype on both 11.1 and 11.3, while it still works on Kubuntu. When I unmuted Front Mic on 11.1 I've suddenly found that it actually enabled Line In. I think now that something went bad with card detecting. Probably on 11.1 with latest git versions my card is not properly detected and some "defaults" are used. I'm attaching here 2 outputs of alsa-info.sh - 11.1_stable captured with alsa-driver compiled from stable 1.0.23 sources. 11.1_git25 captured with driver compiled with git20100925 version from opensuse/multimedia: repository. For me the problem with 11.1 is resolved, but probably it's worth checking why latest version from git behave this way. I'll continue with 11.3 investigation later on. Regards, -- Mark Goldstein

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[alsa-devel] [PATCH] ASoC: Accept any logical value for WM8962 GPIO set()
by Mark Brown 31 Jan '11

31 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5c7b730..3b71dd6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, - WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT); + WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT); } static int wm8962_gpio_direction_out(struct gpio_chip *chip, -- 1.7.2.3

2 1

[PATCH] OMAP4: hwmod data: Add McBSP
by Benoit Cousson 31 Jan '11

31 Jan '11

Add mcbsp data including a revision member in hwmod_class in order to provide mcbsp revision information for different omap. Signed-off-by: Benoit Cousson <b-cousson(a)ti.com> Signed-off-by: Kishon Vijay Abraham I <kishon(a)ti.com> Signed-off-by: Charulatha V <charu(a)ti.com> [b-cousson(a)ti.com: Remove the mcbsp4 memory name, re-order properly the structures, update changelog] --- arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 301 +++++++++++++++++++++++++++- arch/arm/plat-omap/include/plat/mcbsp.h | 1 + 2 files changed, 298 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c index 4ff4eff..0616fb2 100644 --- a/arch/arm/mach-omap2/omap_hwmod_44xx_data.c +++ b/arch/arm/mach-omap2/omap_hwmod_44xx_data.c @@ -26,6 +26,7 @@ #include <plat/dma.h> #include <plat/mcspi.h> #include <plat/dmtimer.h> +#include <plat/mcbsp.h> #include "omap_hwmod_common_data.h" @@ -534,10 +535,6 @@ static struct omap_hwmod omap44xx_mpu_private_hwmod = { * kbd * mailbox * mcasp - * mcbsp1 - * mcbsp2 - * mcbsp3 - * mcbsp4 * mcpdm * mmc1 * mmc2 @@ -2008,6 +2005,296 @@ static struct omap_hwmod omap44xx_iva_hwmod = { }; /* + * 'mcbsp' class + * multi channel buffered serial port controller + */ + +static struct omap_hwmod_class_sysconfig omap44xx_mcbsp_sysc = { + .sysc_offs = 0x008c, + .sysc_flags = (SYSC_HAS_CLOCKACTIVITY | SYSC_HAS_ENAWAKEUP | + SYSC_HAS_SIDLEMODE | SYSC_HAS_SOFTRESET), + .idlemodes = (SIDLE_FORCE | SIDLE_NO | SIDLE_SMART), + .sysc_fields = &omap_hwmod_sysc_type1, +}; + +static struct omap_hwmod_class omap44xx_mcbsp_hwmod_class = { + .name = "mcbsp", + .sysc = &omap44xx_mcbsp_sysc, + .rev = MCBSP_CONFIG_TYPE4, +}; + +/* mcbsp1 */ +static struct omap_hwmod omap44xx_mcbsp1_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp1_irqs[] = { + { .irq = 17 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp1_sdma_reqs[] = { + { .name = "tx", .dma_req = 32 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 33 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp1_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40122000, + .pa_end = 0x401220ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp1 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp1 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp1_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp1_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp1_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp1_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49022000, + .pa_end = 0x490220ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp1 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp1_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp1_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp1_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp1_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp1 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp1_slaves[] = { + &omap44xx_l4_abe__mcbsp1, + &omap44xx_l4_abe__mcbsp1_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp1_hwmod = { + .name = "mcbsp1", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp1_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp1_irqs), + .sdma_reqs = omap44xx_mcbsp1_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp1_sdma_reqs), + .main_clk = "mcbsp1_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP1_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp1_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp1_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp2 */ +static struct omap_hwmod omap44xx_mcbsp2_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp2_irqs[] = { + { .irq = 22 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp2_sdma_reqs[] = { + { .name = "tx", .dma_req = 16 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 17 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp2_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40124000, + .pa_end = 0x401240ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp2 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp2 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp2_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp2_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp2_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp2_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49024000, + .pa_end = 0x490240ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp2 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp2_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp2_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp2_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp2_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp2 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp2_slaves[] = { + &omap44xx_l4_abe__mcbsp2, + &omap44xx_l4_abe__mcbsp2_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp2_hwmod = { + .name = "mcbsp2", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp2_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp2_irqs), + .sdma_reqs = omap44xx_mcbsp2_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp2_sdma_reqs), + .main_clk = "mcbsp2_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP2_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp2_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp2_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp3 */ +static struct omap_hwmod omap44xx_mcbsp3_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp3_irqs[] = { + { .irq = 23 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp3_sdma_reqs[] = { + { .name = "tx", .dma_req = 18 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 19 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp3_addrs[] = { + { + .name = "mpu", + .pa_start = 0x40126000, + .pa_end = 0x401260ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp3 */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp3 = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp3_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp3_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp3_addrs), + .user = OCP_USER_MPU, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp3_dma_addrs[] = { + { + .name = "dma", + .pa_start = 0x49026000, + .pa_end = 0x490260ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_abe -> mcbsp3 (dma) */ +static struct omap_hwmod_ocp_if omap44xx_l4_abe__mcbsp3_dma = { + .master = &omap44xx_l4_abe_hwmod, + .slave = &omap44xx_mcbsp3_hwmod, + .clk = "ocp_abe_iclk", + .addr = omap44xx_mcbsp3_dma_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp3_dma_addrs), + .user = OCP_USER_SDMA, +}; + +/* mcbsp3 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp3_slaves[] = { + &omap44xx_l4_abe__mcbsp3, + &omap44xx_l4_abe__mcbsp3_dma, +}; + +static struct omap_hwmod omap44xx_mcbsp3_hwmod = { + .name = "mcbsp3", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp3_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp3_irqs), + .sdma_reqs = omap44xx_mcbsp3_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp3_sdma_reqs), + .main_clk = "mcbsp3_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM1_ABE_MCBSP3_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp3_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp3_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* mcbsp4 */ +static struct omap_hwmod omap44xx_mcbsp4_hwmod; +static struct omap_hwmod_irq_info omap44xx_mcbsp4_irqs[] = { + { .irq = 16 + OMAP44XX_IRQ_GIC_START }, +}; + +static struct omap_hwmod_dma_info omap44xx_mcbsp4_sdma_reqs[] = { + { .name = "tx", .dma_req = 30 + OMAP44XX_DMA_REQ_START }, + { .name = "rx", .dma_req = 31 + OMAP44XX_DMA_REQ_START }, +}; + +static struct omap_hwmod_addr_space omap44xx_mcbsp4_addrs[] = { + { + .pa_start = 0x48096000, + .pa_end = 0x480960ff, + .flags = ADDR_TYPE_RT + }, +}; + +/* l4_per -> mcbsp4 */ +static struct omap_hwmod_ocp_if omap44xx_l4_per__mcbsp4 = { + .master = &omap44xx_l4_per_hwmod, + .slave = &omap44xx_mcbsp4_hwmod, + .clk = "l4_div_ck", + .addr = omap44xx_mcbsp4_addrs, + .addr_cnt = ARRAY_SIZE(omap44xx_mcbsp4_addrs), + .user = OCP_USER_MPU | OCP_USER_SDMA, +}; + +/* mcbsp4 slave ports */ +static struct omap_hwmod_ocp_if *omap44xx_mcbsp4_slaves[] = { + &omap44xx_l4_per__mcbsp4, +}; + +static struct omap_hwmod omap44xx_mcbsp4_hwmod = { + .name = "mcbsp4", + .class = &omap44xx_mcbsp_hwmod_class, + .mpu_irqs = omap44xx_mcbsp4_irqs, + .mpu_irqs_cnt = ARRAY_SIZE(omap44xx_mcbsp4_irqs), + .sdma_reqs = omap44xx_mcbsp4_sdma_reqs, + .sdma_reqs_cnt = ARRAY_SIZE(omap44xx_mcbsp4_sdma_reqs), + .main_clk = "mcbsp4_fck", + .prcm = { + .omap4 = { + .clkctrl_reg = OMAP4430_CM_L4PER_MCBSP4_CLKCTRL, + }, + }, + .slaves = omap44xx_mcbsp4_slaves, + .slaves_cnt = ARRAY_SIZE(omap44xx_mcbsp4_slaves), + .omap_chip = OMAP_CHIP_INIT(CHIP_IS_OMAP4430), +}; + +/* * 'mcspi' class * multichannel serial port interface (mcspi) / master/slave synchronous serial * bus @@ -3568,6 +3855,12 @@ static __initdata struct omap_hwmod *omap44xx_hwmods[] = { &omap44xx_iva_seq0_hwmod, &omap44xx_iva_seq1_hwmod, + /* mcbsp class */ + &omap44xx_mcbsp1_hwmod, + &omap44xx_mcbsp2_hwmod, + &omap44xx_mcbsp3_hwmod, + &omap44xx_mcbsp4_hwmod, + /* mcspi class */ &omap44xx_mcspi1_hwmod, &omap44xx_mcspi2_hwmod, diff --git a/arch/arm/plat-omap/include/plat/mcbsp.h b/arch/arm/plat-omap/include/plat/mcbsp.h index 8d8434b..06bd1dc 100644 --- a/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/arch/arm/plat-omap/include/plat/mcbsp.h @@ -39,6 +39,7 @@ static struct platform_device omap_mcbsp##port_nr = { \ #define MCBSP_CONFIG_TYPE2 0x2 #define MCBSP_CONFIG_TYPE3 0x3 +#define MCBSP_CONFIG_TYPE4 0x4 #define OMAP7XX_MCBSP1_BASE 0xfffb1000 #define OMAP7XX_MCBSP2_BASE 0xfffb1800 -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Accept any logical value WM8903 GPIO set()
by Mark Brown 31 Jan '11

31 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 54383c3..27cd8f5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1904,7 +1904,8 @@ static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8903->codec; snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { -- 1.7.2.3

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[alsa-devel] [PATCH] soc: neo1973_wm8753 audio support does not require scoop
by H Hartley Sweeten 31 Jan '11

31 Jan '11

This driver does not use any of the functionality provided by the scoop hardware. Remove the unneeded header. Signed-off-by: H Hartley Sweeten <hsweeten(a)visionengravers.com> --- diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c7a2451..c5b3113 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -24,7 +24,6 @@ #include <sound/tlv.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h>

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[alsa-devel] Hello, first mail and a quesion.
by Matías Goinheix Freitas 31 Jan '11

31 Jan '11

Hello people, This is my first mail here. Before purchasing the cheap awful USB interface I have now, I was in love with the Saffire 6 USB. Sadly it didn't run on linux. A friend of mine got it (he uses another OS) and so I tested for myself that it's not recognized by Ubuntu 10.04. My question is this: I have access to that interface because I can borrow it anytime I want. I don't have the slightest idea on developing a driver or hacking stuff (I suscribed to this mailing list because I thought it's the one I could have contact with developers even though I'm not, if I'm in the wrong place I will unsuscribe it's not my intention to bother). Is there any way I could help developers interested in creating a driver for this interfase? I don't know if that makes sense, but maybe there's a project going on that I'm unaware of and they could use some info about that interface. I could copy/paste/send them the info they needed, or run the commands they tell me to run and give them the info. Thank you, Matías

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[alsa-devel] [PATCH 1/7] ASoC: Move card list initialization to snd_soc_register_card
by Stephen Warren 31 Jan '11

31 Jan '11

All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was only called for cards backed by a "soc-audio" platform device, via soc_probe(). However, it's also needed for cards backed by other platform devices, and registered directly via snd_soc_register_card(). Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9dfbb8f..3feddd9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1881,7 +1881,6 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { @@ -3122,6 +3121,8 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + snd_soc_initialize_card_lists(card); + soc_init_card_debugfs(card); card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * -- 1.7.1

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