From 92429069d0fc9f52d436c9067c5b5c392e3f8876 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 19 Mar 2009 09:32:01 +0100 Subject: ASoC: pxa-ssp: Use 16-bit DMA for magician stereo Now magician and similar boards can use network mode with only one active slot to explicitly set 16 bit frame width, even for S16_LE stereo sound. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b0bf40973d5..c7c1996a544 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); + int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; /* select correct DMA params */ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) dma = 1; /* capture DMA offset is 1,3 */ - if (chn == 2) - dma += 2; /* stereo DMA offset is 2, mono is 0 */ + /* Network mode with one active slot (ttsa == 1) can be used + * to force 16-bit frame width on the wire (for S16_LE), even + * with two channels. Use 16-bit DMA transfers for this case. + */ + if (((chn == 2) && (ttsa != 1)) || (width == 32)) + dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); @@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !ttsa) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- cgit v1.2.3 From 7377226c344a7295a7573dce400ce9ddd42f0ca4 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 19 Mar 2009 09:34:46 +0100 Subject: ASoC: Add Magician machine support HTC Magician has a Philips UDA1380 codec connected via SSP1 (playback) and I2S (capture). There is a flip-flop between the SSP frame clock output and the codec's word select input pin. To make the codec see proper I2S input, the SSP has to send two frames per sample. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 10 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/magician.c | 560 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 572 insertions(+) create mode 100644 sound/soc/pxa/magician.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 5998ab366e8..ad8a10fe629 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_PXA2XX_SOC_MAGICIAN + tristate "SoC Audio support for HTC Magician" + depends on SND_PXA2XX_SOC && MACH_MAGICIAN + select SND_PXA2XX_SOC_I2S + select SND_PXA_SOC_SSP + select SND_SOC_UDA1380 + help + Say Y if you want to add support for SoC audio on the + HTC Magician. + config SND_PXA2XX_SOC_MIOA701 tristate "SoC Audio support for MIO A701" depends on SND_PXA2XX_SOC && MACH_MIOA701 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 8ed881c5e5c..4b90c3ccae4 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o @@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c new file mode 100644 index 00000000000..f7c4544f785 --- /dev/null +++ b/sound/soc/pxa/magician.c @@ -0,0 +1,560 @@ +/* + * SoC audio for HTC Magician + * + * Copyright (c) 2006 Philipp Zabel + * + * based on spitz.c, + * Authors: Liam Girdwood + * Richard Purdie + * + * 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 +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include "../codecs/uda1380.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-i2s.h" +#include "pxa-ssp.h" + +#define MAGICIAN_MIC 0 +#define MAGICIAN_MIC_EXT 1 + +static int magician_hp_switch; +static int magician_spk_switch = 1; +static int magician_in_sel = MAGICIAN_MIC; + +static void magician_ext_control(struct snd_soc_codec *codec) +{ + if (magician_spk_switch) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + if (magician_hp_switch) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + switch (magician_in_sel) { + case MAGICIAN_MIC: + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case MAGICIAN_MIC_EXT: + snd_soc_dapm_disable_pin(codec, "Call Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + break; + } + + snd_soc_dapm_sync(codec); +} + +static int magician_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->card->codec; + + /* check the jack status at stream startup */ + magician_ext_control(codec); + + return 0; +} + +/* + * Magician uses SSP port for playback. + */ +static int magician_playback_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; + unsigned int acps, acds, width, rate; + unsigned int div4 = PXA_SSP_CLK_SCDB_4; + int ret = 0; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + + /* + * rate = SSPSCLK / (2 * width(16 or 32)) + * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) + */ + switch (params_rate(params)) { + case 8000: + /* off by a factor of 2: bug in the PXA27x audio clock? */ + acps = 32842000; + switch (width) { + case 16: + /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_16; + break; + case 32: + /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_8; + } + break; + case 11025: + acps = 5622000; + switch (width) { + case 16: + /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_4; + break; + case 32: + /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + } + break; + case 22050: + acps = 5622000; + switch (width) { + case 16: + /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 44100: + acps = 5622000; + switch (width) { + case 16: + /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 48000: + acps = 12235000; + switch (width) { + case 16: + /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + break; + case 32: + /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + } + break; + case 96000: + acps = 12235000; + switch (width) { + case 16: + /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_1; + break; + case 32: + /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ + acds = PXA_SSP_CLK_AUDIO_DIV_2; + div4 = PXA_SSP_CLK_SCDB_1; + break; + } + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | + 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_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + if (ret < 0) + return ret; + + /* set audio clock as clock source */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* set the SSP audio system clock ACDS divider */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + /* set the SSP audio system clock SCDB divider4 */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + PXA_SSP_AUDIO_DIV_SCDB, div4); + if (ret < 0) + return ret; + + /* set SSP audio pll clock */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Magician uses I2S for capture. + */ +static int magician_capture_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; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | 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_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops magician_capture_ops = { + .startup = magician_startup, + .hw_params = magician_capture_hw_params, +}; + +static struct snd_soc_ops magician_playback_ops = { + .startup = magician_startup, + .hw_params = magician_playback_hw_params, +}; + +static int magician_get_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_hp_switch; + return 0; +} + +static int magician_set_hp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_hp_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_hp_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_spk_switch; + return 0; +} + +static int magician_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (magician_spk_switch == ucontrol->value.integer.value[0]) + return 0; + + magician_spk_switch = ucontrol->value.integer.value[0]; + magician_ext_control(codec); + return 1; +} + +static int magician_get_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = magician_in_sel; + return 0; +} + +static int magician_set_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (magician_in_sel == ucontrol->value.integer.value[0]) + return 0; + + magician_in_sel = ucontrol->value.integer.value[0]; + + switch (magician_in_sel) { + case MAGICIAN_MIC: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1); + break; + case MAGICIAN_MIC_EXT: + gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0); + } + + return 1; +} + +static int magician_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int magician_mic_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* magician machine dapm widgets */ +static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power), + SND_SOC_DAPM_SPK("Speaker", magician_spk_power), + SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias), + SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias), +}; + +/* magician machine audio_map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone connected to VOUTL, VOUTR */ + {"Headphone Jack", NULL, "VOUTL"}, + {"Headphone Jack", NULL, "VOUTR"}, + + /* Speaker connected to VOUTL, VOUTR */ + {"Speaker", NULL, "VOUTL"}, + {"Speaker", NULL, "VOUTR"}, + + /* Mics are connected to VINM */ + {"VINM", NULL, "Headset Mic"}, + {"VINM", NULL, "Call Mic"}, +}; + +static const char *input_select[] = {"Call Mic", "Headset Mic"}; +static const struct soc_enum magician_in_sel_enum = + SOC_ENUM_SINGLE_EXT(2, input_select); + +static const struct snd_kcontrol_new uda1380_magician_controls[] = { + SOC_SINGLE_BOOL_EXT("Headphone Switch", + (unsigned long)&magician_hp_switch, + magician_get_hp, magician_set_hp), + SOC_SINGLE_BOOL_EXT("Speaker Switch", + (unsigned long)&magician_spk_switch, + magician_get_spk, magician_set_spk), + SOC_ENUM_EXT("Input Select", magician_in_sel_enum, + magician_get_input, magician_set_input), +}; + +/* + * Logic for a uda1380 as connected on a HTC Magician + */ +static int magician_uda1380_init(struct snd_soc_codec *codec) +{ + int err; + + /* NC codec pins */ + snd_soc_dapm_nc_pin(codec, "VOUTLHP"); + snd_soc_dapm_nc_pin(codec, "VOUTRHP"); + + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(codec, "VINL"); + snd_soc_dapm_nc_pin(codec, "VINR"); + + /* Add magician specific controls */ + err = snd_soc_add_controls(codec, uda1380_magician_controls, + ARRAY_SIZE(uda1380_magician_controls)); + if (err < 0) + return err; + + /* Add magician specific widgets */ + snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets, + ARRAY_SIZE(uda1380_dapm_widgets)); + + /* Set up magician specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* magician digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link magician_dai[] = { +{ + .name = "uda1380", + .stream_name = "UDA1380 Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], + .init = magician_uda1380_init, + .ops = &magician_playback_ops, +}, +{ + .name = "uda1380", + .stream_name = "UDA1380 Capture", + .cpu_dai = &pxa_i2s_dai, + .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE], + .ops = &magician_capture_ops, +} +}; + +/* magician audio machine driver */ +static struct snd_soc_card snd_soc_card_magician = { + .name = "Magician", + .dai_link = magician_dai, + .num_links = ARRAY_SIZE(magician_dai), + .platform = &pxa2xx_soc_platform, +}; + +/* magician audio private data */ +static struct uda1380_setup_data magician_uda1380_setup = { + .i2c_address = 0x18, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +/* magician audio subsystem */ +static struct snd_soc_device magician_snd_devdata = { + .card = &snd_soc_card_magician, + .codec_dev = &soc_codec_dev_uda1380, + .codec_data = &magician_uda1380_setup, +}; + +static struct platform_device *magician_snd_device; + +static int __init magician_init(void) +{ + int ret; + + if (!machine_is_magician()) + return -ENODEV; + + ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); + if (ret) + goto err_request_power; + ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); + if (ret) + goto err_request_reset; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); + if (ret) + goto err_request_spk; + ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER"); + if (ret) + goto err_request_ep; + ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER"); + if (ret) + goto err_request_mic; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0"); + if (ret) + goto err_request_in_sel0; + ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1"); + if (ret) + goto err_request_in_sel1; + + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); + gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); + + /* we may need to have the clock running here - pH5 */ + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); + udelay(5); + gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); + + magician_snd_device = platform_device_alloc("soc-audio", -1); + if (!magician_snd_device) { + ret = -ENOMEM; + goto err_pdev; + } + + platform_set_drvdata(magician_snd_device, &magician_snd_devdata); + magician_snd_devdata.dev = &magician_snd_device->dev; + ret = platform_device_add(magician_snd_device); + if (ret) { + platform_device_put(magician_snd_device); + goto err_pdev; + } + + return 0; + +err_pdev: + gpio_free(EGPIO_MAGICIAN_IN_SEL1); +err_request_in_sel1: + gpio_free(EGPIO_MAGICIAN_IN_SEL0); +err_request_in_sel0: + gpio_free(EGPIO_MAGICIAN_MIC_POWER); +err_request_mic: + gpio_free(EGPIO_MAGICIAN_EP_POWER); +err_request_ep: + gpio_free(EGPIO_MAGICIAN_SPK_POWER); +err_request_spk: + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); +err_request_reset: + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +err_request_power: + return ret; +} + +static void __exit magician_exit(void) +{ + platform_device_unregister(magician_snd_device); + + gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); + gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); + + gpio_free(EGPIO_MAGICIAN_IN_SEL1); + gpio_free(EGPIO_MAGICIAN_IN_SEL0); + gpio_free(EGPIO_MAGICIAN_MIC_POWER); + gpio_free(EGPIO_MAGICIAN_EP_POWER); + gpio_free(EGPIO_MAGICIAN_SPK_POWER); + gpio_free(EGPIO_MAGICIAN_CODEC_RESET); + gpio_free(EGPIO_MAGICIAN_CODEC_POWER); +} + +module_init(magician_init); +module_exit(magician_exit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC Magician"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From a4d11fe50c238a7da5225d1399314c3505cbd792 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Wed, 25 Mar 2009 18:20:37 -0500 Subject: ASoC: remove trigger delay in Freescale MPC8610 sound driver Remove the delay from the trigger function in the Freescale MPC8610 sound driver when capture is started. This delay was used to ensure that the DMA controller was active when ALSA call the .pointer function to request a DMA transfer status. A better approach is for the .pointer function to detect that DMA has not started, and return zero instead. This change eliminates the need for the delay. Also add some related code to check for a DMA programming error, and report XRUN if it occurs. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 17 +++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 20 ++------------------ 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb8570cd7..2c4892c853c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + /* + * When capture is started, the SSI immediately starts to fill its FIFO. + * This means that the DMA controller is not started until the FIFO is + * full. However, ALSA calls this function before that happens, when + * MR.DAR is still zero. In this case, just return zero to indicate + * that nothing has been received yet. + */ + if (!position) + return 0; + + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "dma pointer is out of range, halting stream\n"); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 169bca295b7..72823a2b33d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -466,28 +466,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - } else { - long timeout = jiffies + 10; - + else setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - - /* Wait until the SSI has filled its FIFO. Without this - * delay, ALSA complains about overruns. When the FIFO - * is full, the DMA controller initiates its first - * transfer. Until then, however, the DMA's DAR - * register is zero, which translates to an - * out-of-bounds pointer. This makes ALSA think an - * overrun has occurred. - */ - while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && - (jiffies < timeout)); - if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) - return -EIO; - } break; case SNDRV_PCM_TRIGGER_STOP: -- cgit v1.2.3 From 057de50c0d34b4ef7e15b7a8442a36a396d99c00 Mon Sep 17 00:00:00 2001 From: Luotao Fu Date: Thu, 26 Mar 2009 13:18:03 +0100 Subject: pxa2xx-ac97: fix displaying GSR after reset timeout the variable gsr_bit is set in isr. It is however set to 0 and interrupts are disabled prior to reset. Hence it doesn't make a lot of sense to show the content of gsr_bit in case of a reset timeout. Signed-off-by: Luotao Fu Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 2e6355f4cbb..71bef45e9d3 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -239,6 +239,8 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); @@ -255,10 +257,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); - - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } @@ -269,6 +271,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { + unsigned long gsr; + #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); @@ -286,9 +290,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); - if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { + gsr = GSR | gsr_bits; + if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, gsr); return false; } -- cgit v1.2.3 From d5a908b27adfd7e67b5ab98f674892badcca19c6 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 26 Mar 2009 11:42:38 -0500 Subject: ASoC: trim SSI sysfs statistics in Freescale MPC8610 sound drivers Optimize the display of SSI statistics in the Freescale MPC8610 sound driver to display the status count only of the interrupts that were actually enabled. Previously, it would display the counts of all SISR status bits, even those that were not enabled. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 79 +++++++++++++++++++++++++++++-------------------- 1 file changed, 47 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 72823a2b33d..3711d8454d9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -60,6 +60,13 @@ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE) #endif +/* SIER bitflag of interrupts to enable */ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ + CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ + CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ + CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ + CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) + /** * fsl_ssi_private: per-SSI private data * @@ -140,7 +147,7 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) were interrupted for. We mask it with the Interrupt Enable register so that we only check for events that we're interested in. */ - sisr = in_be32(&ssi->sisr) & in_be32(&ssi->sier); + sisr = in_be32(&ssi->sisr) & SIER_FLAGS; if (sisr & CCSR_SSI_SISR_RFRC) { ssi_private->stats.rfrc++; @@ -324,12 +331,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, */ /* 4. Enable the interrupts and DMA requests */ - out_be32(&ssi->sier, - CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | - CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN); + out_be32(&ssi->sier, SIER_FLAGS); /* * Set the watermark for transmit FIFI 0 and receive FIFO 0. We @@ -590,39 +592,52 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .ops = &fsl_ssi_dai_ops, }; +/* Show the statistics of a flag only if its interrupt is enabled. The + * compiler will optimze this code to a no-op if the interrupt is not + * enabled. + */ +#define SIER_SHOW(flag, name) \ + do { \ + if (SIER_FLAGS & CCSR_SSI_SIER_##flag) \ + length += sprintf(buf + length, #name "=%u\n", \ + ssi_private->stats.name); \ + } while (0) + + /** * fsl_sysfs_ssi_show: display SSI statistics * - * Display the statistics for the current SSI device. + * Display the statistics for the current SSI device. To avoid confusion, + * we only show those counts that are enabled. */ static ssize_t fsl_sysfs_ssi_show(struct device *dev, struct device_attribute *attr, char *buf) { struct fsl_ssi_private *ssi_private = - container_of(attr, struct fsl_ssi_private, dev_attr); - ssize_t length; - - length = sprintf(buf, "rfrc=%u", ssi_private->stats.rfrc); - length += sprintf(buf + length, "\ttfrc=%u", ssi_private->stats.tfrc); - length += sprintf(buf + length, "\tcmdau=%u", ssi_private->stats.cmdau); - length += sprintf(buf + length, "\tcmddu=%u", ssi_private->stats.cmddu); - length += sprintf(buf + length, "\trxt=%u", ssi_private->stats.rxt); - length += sprintf(buf + length, "\trdr1=%u", ssi_private->stats.rdr1); - length += sprintf(buf + length, "\trdr0=%u", ssi_private->stats.rdr0); - length += sprintf(buf + length, "\ttde1=%u", ssi_private->stats.tde1); - length += sprintf(buf + length, "\ttde0=%u", ssi_private->stats.tde0); - length += sprintf(buf + length, "\troe1=%u", ssi_private->stats.roe1); - length += sprintf(buf + length, "\troe0=%u", ssi_private->stats.roe0); - length += sprintf(buf + length, "\ttue1=%u", ssi_private->stats.tue1); - length += sprintf(buf + length, "\ttue0=%u", ssi_private->stats.tue0); - length += sprintf(buf + length, "\ttfs=%u", ssi_private->stats.tfs); - length += sprintf(buf + length, "\trfs=%u", ssi_private->stats.rfs); - length += sprintf(buf + length, "\ttls=%u", ssi_private->stats.tls); - length += sprintf(buf + length, "\trls=%u", ssi_private->stats.rls); - length += sprintf(buf + length, "\trff1=%u", ssi_private->stats.rff1); - length += sprintf(buf + length, "\trff0=%u", ssi_private->stats.rff0); - length += sprintf(buf + length, "\ttfe1=%u", ssi_private->stats.tfe1); - length += sprintf(buf + length, "\ttfe0=%u\n", ssi_private->stats.tfe0); + container_of(attr, struct fsl_ssi_private, dev_attr); + ssize_t length = 0; + + SIER_SHOW(RFRC_EN, rfrc); + SIER_SHOW(TFRC_EN, tfrc); + SIER_SHOW(CMDAU_EN, cmdau); + SIER_SHOW(CMDDU_EN, cmddu); + SIER_SHOW(RXT_EN, rxt); + SIER_SHOW(RDR1_EN, rdr1); + SIER_SHOW(RDR0_EN, rdr0); + SIER_SHOW(TDE1_EN, tde1); + SIER_SHOW(TDE0_EN, tde0); + SIER_SHOW(ROE1_EN, roe1); + SIER_SHOW(ROE0_EN, roe0); + SIER_SHOW(TUE1_EN, tue1); + SIER_SHOW(TUE0_EN, tue0); + SIER_SHOW(TFS_EN, tfs); + SIER_SHOW(RFS_EN, rfs); + SIER_SHOW(TLS_EN, tls); + SIER_SHOW(RLS_EN, rls); + SIER_SHOW(RFF1_EN, rff1); + SIER_SHOW(RFF0_EN, rff0); + SIER_SHOW(TFE1_EN, tfe1); + SIER_SHOW(TFE0_EN, tfe0); return length; } -- cgit v1.2.3 From 31ad0f31c3a45ba489203eef7e71d3215005afbc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 27 Mar 2009 10:39:07 +0200 Subject: ASoC: TWL4030: 96KHz playback support TWL4030 supports 96KHz sample playback, but only playback. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 97738e2ece0..b07d8d68a93 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1395,7 +1395,7 @@ struct snd_soc_dai twl4030_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = TWL4030_RATES, + .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, .formats = TWL4030_FORMATS,}, .capture = { .stream_name = "Capture", -- cgit v1.2.3 From 7220b9f4bd4fad41f6f7299fe74c2c38ec85d793 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 27 Mar 2009 10:39:08 +0200 Subject: ASoC: TWL4030: Add constrains for second stream In case of duplex mode (capture and playback at the same time), the second stream has to have the same parameters (rate, sample size) as the already running stream. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 54 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b07d8d68a93..4199498a891 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -122,6 +122,9 @@ struct twl4030_priv { unsigned int bypass_state; unsigned int codec_powered; unsigned int codec_muted; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; }; /* @@ -1217,6 +1220,50 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int twl4030_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + /* If we already have a playback or capture going then constrain + * this substream to match it. + */ + if (twl4030->master_substream) { + struct snd_pcm_runtime *master_runtime; + master_runtime = twl4030->master_substream->runtime; + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + twl4030->slave_substream = substream; + } else + twl4030->master_substream = substream; + + return 0; +} + +static void twl4030_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct twl4030_priv *twl4030 = codec->private_data; + + if (twl4030->master_substream == substream) + twl4030->master_substream = twl4030->slave_substream; + + twl4030->slave_substream = NULL; +} + static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1224,8 +1271,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; + if (substream == twl4030->slave_substream) + /* Ignoring hw_params for slave substream */ + return 0; + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1384,6 +1436,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) static struct snd_soc_dai_ops twl4030_dai_ops = { + .startup = twl4030_startup, + .shutdown = twl4030_shutdown, .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, -- cgit v1.2.3 From 6984992bf0520a07b931124d33f46b46437f6e1c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 27 Mar 2009 15:32:01 +0200 Subject: ASoC: OMAP: Set minimum buffer size constraint for McBSP2 in OMAP3 McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During initial playback startup, this FIFO is keeping the DMA request active until the FIFO is full. So now if ALSA buffer size is smaller, DMA is looping around it while filling up the HW FIFO, generating burst of interrupts as well and SW doesn't have any change to fill enough data. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d6882be3345..9c09b94f0cf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,6 +146,17 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; + if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) { + /* + * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. + * Set constraint for minimum buffer size to the same than FIFO + * size in order to avoid underruns in playback startup because + * HW is keeping the DMA request active until FIFO is filled. + */ + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX); + } + if (!cpu_dai->active) err = omap_mcbsp_request(mcbsp_data->bus_id); -- cgit v1.2.3 From a7808331f1ea6c7f89a14d1d94eafc62615b997b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 27 Mar 2009 17:14:52 +0000 Subject: ASoC: Add some documentation for the ASoC jack API A brief overview of how the components of the API fit together. Signed-off-by: Mark Brown --- Documentation/sound/alsa/soc/jack.txt | 71 +++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) create mode 100644 Documentation/sound/alsa/soc/jack.txt diff --git a/Documentation/sound/alsa/soc/jack.txt b/Documentation/sound/alsa/soc/jack.txt new file mode 100644 index 00000000000..fcf82a41729 --- /dev/null +++ b/Documentation/sound/alsa/soc/jack.txt @@ -0,0 +1,71 @@ +ASoC jack detection +=================== + +ALSA has a standard API for representing physical jacks to user space, +the kernel side of which can be seen in include/sound/jack.h. ASoC +provides a version of this API adding two additional features: + + - It allows more than one jack detection method to work together on one + user visible jack. In embedded systems it is common for multiple + to be present on a single jack but handled by separate bits of + hardware. + + - Integration with DAPM, allowing DAPM endpoints to be updated + automatically based on the detected jack status (eg, turning off the + headphone outputs if no headphones are present). + +This is done by splitting the jacks up into three things working +together: the jack itself represented by a struct snd_soc_jack, sets of +snd_soc_jack_pins representing DAPM endpoints to update and blocks of +code providing jack reporting mechanisms. + +For example, a system may have a stereo headset jack with two reporting +mechanisms, one for the headphone and one for the microphone. Some +systems won't be able to use their speaker output while a headphone is +connected and so will want to make sure to update both speaker and +headphone when the headphone jack status changes. + +The jack - struct snd_soc_jack +============================== + +This represents a physical jack on the system and is what is visible to +user space. The jack itself is completely passive, it is set up by the +machine driver and updated by jack detection methods. + +Jacks are created by the machine driver calling snd_soc_jack_new(). + +snd_soc_jack_pin +================ + +These represent a DAPM pin to update depending on some of the status +bits supported by the jack. Each snd_soc_jack has zero or more of these +which are updated automatically. They are created by the machine driver +and associated with the jack using snd_soc_jack_add_pins(). The status +of the endpoint may configured to be the opposite of the jack status if +required (eg, enabling a built in microphone if a microphone is not +connected via a jack). + +Jack detection methods +====================== + +Actual jack detection is done by code which is able to monitor some +input to the system and update a jack by calling snd_soc_jack_report(), +specifying a subset of bits to update. The jack detection code should +be set up by the machine driver, taking configuration for the jack to +update and the set of things to report when the jack is connected. + +Often this is done based on the status of a GPIO - a handler for this is +provided by the snd_soc_jack_add_gpio() function. Other methods are +also available, for example integrated into CODECs. One example of +CODEC integrated jack detection can be see in the WM8350 driver. + +Each jack may have multiple reporting mechanisms, though it will need at +least one to be useful. + +Machine drivers +=============== + +These are all hooked together by the machine driver depending on the +system hardware. The machine driver will set up the snd_soc_jack and +the list of pins to update then set up one or more jack detection +mechanisms to update that jack based on their current status. -- cgit v1.2.3 From 64ab9baa00fa99070da993f00173c35a8e99abfa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Mar 2009 11:27:03 +0100 Subject: ASoC: Don't defer resume work for AC97 codecs AC97 devices may have other drivers hanging off them directly so need to have resumed when the resume function returns meaning that we can't defer the resume - complete it immediately for them. Non-AC97 devices should not have other drivers hanging directly off the ASoC devices. We only really need the deferral for non-AC97 devices - it's there since some I2C buses are very slow and non-AC97 codecs often have large numbers of registers to restore and require delays to bring the codec up cleanly leading to a substantial impact on overall resume time. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6e710f705a7..6c62d4a54cd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -767,11 +767,21 @@ static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - dev_dbg(socdev->dev, "scheduling resume work\n"); - - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + /* AC97 devices might have other drivers hanging off them so + * need to resume immediately. Other drivers don't have that + * problem and may take a substantial amount of time to resume + * due to I/O costs and anti-pop so handle them out of line. + */ + if (cpu_dai->ac97_control) { + dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(socdev->dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(socdev->dev, "resume work item may be lost\n"); + } return 0; } -- cgit v1.2.3 From 4ac5c61f0fc9b01946911a52d827f67947ab01a8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Apr 2009 19:35:01 +0100 Subject: ASoC: Set parent for AC97 devices we register Ensure that any AC97 devices that bind to the CODEC are below the ASoC device in the device tree so the suspend and resume code can figure out what order to handle them in. Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6c62d4a54cd..99712f652d0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -98,7 +98,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) int err; codec->ac97->dev.bus = &ac97_bus_type; - codec->ac97->dev.parent = NULL; + codec->ac97->dev.parent = codec->card->dev; codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", -- cgit v1.2.3 From 0a11b16853b642a26eb248ac4db422e6dfa04ae5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 2 Apr 2009 15:49:41 +0100 Subject: ASoC: Implement suspend and resume operations for WM9705 Without this the WM9705 driver fails badly when resuming. Tested-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 3265817c5c2..6e23a81dba7 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec) return -EIO; } +#ifdef CONFIG_PM +static int wm9705_soc_suspend(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff); + + return 0; +} + +static int wm9705_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9705_reset(codec); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) { + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + + return 0; +} +#else +#define wm9705_soc_suspend NULL +#define wm9705_soc_resume NULL +#endif + static int wm9705_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev) struct snd_soc_codec_device soc_codec_dev_wm9705 = { .probe = wm9705_soc_probe, .remove = wm9705_soc_remove, + .suspend = wm9705_soc_suspend, + .resume = wm9705_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); -- cgit v1.2.3 From 103f211d0be2bed75b5739de62a10415ef0bbc25 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Apr 2009 14:39:05 +0300 Subject: ASoC: TWL4030: Add actual support for 96KHz playback support Adds the needed code to be able to use 96KHz playback. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 3 +++ sound/soc/codecs/twl4030.h | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4199498a891..bfda7a88e82 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1311,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, case 48000: mode |= TWL4030_APLL_RATE_48000; break; + case 96000: + mode |= TWL4030_APLL_RATE_96000; + break; default: printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", params_rate(params)); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 33dbb144dad..cb63765db1d 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -109,6 +109,7 @@ #define TWL4030_APLL_RATE_32000 0x80 #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_APLL_RATE_96000 0xE0 #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 -- cgit v1.2.3 From 1661c6155589f8faa1338f3cda696ea3f4cb2da1 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Tue, 24 Mar 2009 22:37:14 +0100 Subject: ALSA: opl3sa2: add ZV port control Add ZV port control switch. This patch is done after solution given in the ALSA bug #2872 report. The patch resolves the issue. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opl3sa2.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index ef95279da7a..0481a55334b 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -481,6 +481,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1, OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1), OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1, db_scale_5bit_12db_max), +OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0), }; static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = { -- cgit v1.2.3 From 7d2ac1036b7ff57f73ab64fd897867ddc07bcffe Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sat, 28 Mar 2009 00:27:10 +0100 Subject: ALSA: move snd_powermac's probe function to .devinit.text MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A pointer to snd_pmac_probe is passed to the core via platform_driver_register and so the function must not disappear when the .init sections are discarded. Otherwise (when having HOTPLUG=y) unbinding and binding a device to the driver via sysfs will result in an oops as does a device being registered late. An alternative to this patch is using platform_driver_probe instead of platform_driver_register plus removing the pointer to the probe function from the struct platform_driver. Signed-off-by: Uwe Kleine-König Cc: Jaroslav Kysela Cc: Johannes Berg Cc: Rene Herman Cc: Andrew Morton Signed-off-by: Takashi Iwai --- sound/ppc/powermac.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 5a929069dce..a2b69b8cff4 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -51,7 +51,7 @@ static struct platform_device *device; /* */ -static int __init snd_pmac_probe(struct platform_device *devptr) +static int __devinit snd_pmac_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_pmac *chip; -- cgit v1.2.3 From ff2e7337b5b087620bdea9477f779413a7f096cb Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 1 Apr 2009 14:49:48 -0400 Subject: ALSA: Add 92HD81B1C device id Added device id in struct for codec 92HD81B1C (0x111d76d5). Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b5e108aa8f6..b34d78b88a8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4895,6 +4895,7 @@ again: switch (codec->vendor_id) { case 0x111d7604: case 0x111d7605: + case 0x111d76d5: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; @@ -5707,6 +5708,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, -- cgit v1.2.3 From 8321fc0113a1be0cdfc9cbad1db1de74073acd8f Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:10 +0200 Subject: ALSA: snd-atmel-ac97c: cleanup register definitions This patch will remove traces of channel B registers, since they are not used by the AC97C driver. Channel B might be used for other purposes. The driver also adds channel status bits TXEMPTY and OVRUN and a AC97C_CH_MASK macro to ease clearing a channel settings. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.h | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h index c17bd582598..ecbba5021c8 100644 --- a/sound/atmel/ac97c.h +++ b/sound/atmel/ac97c.h @@ -1,5 +1,5 @@ /* - * Register definitions for the Atmel AC97C controller + * Register definitions for Atmel AC97C * * Copyright (C) 2005-2009 Atmel Corporation * @@ -17,10 +17,6 @@ #define AC97C_CATHR 0x24 #define AC97C_CASR 0x28 #define AC97C_CAMR 0x2c -#define AC97C_CBRHR 0x30 -#define AC97C_CBTHR 0x34 -#define AC97C_CBSR 0x38 -#define AC97C_CBMR 0x3c #define AC97C_CORHR 0x40 #define AC97C_COTHR 0x44 #define AC97C_COSR 0x48 @@ -46,8 +42,10 @@ #define AC97C_MR_VRA (1 << 2) #define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_TXEMPTY (1 << 1) #define AC97C_CSR_UNRUN (1 << 2) #define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_OVRUN (1 << 5) #define AC97C_CSR_ENDTX (1 << 10) #define AC97C_CSR_ENDRX (1 << 14) @@ -61,11 +59,15 @@ #define AC97C_CMR_DMAEN (1 << 22) #define AC97C_SR_CAEVT (1 << 3) +#define AC97C_SR_COEVT (1 << 2) +#define AC97C_SR_WKUP (1 << 1) +#define AC97C_SR_SOF (1 << 0) +#define AC97C_CH_MASK(slot) \ + (0x7 << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CH_ASSIGN(slot, channel) \ (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) #define AC97C_CHANNEL_NONE 0x0 #define AC97C_CHANNEL_A 0x1 -#define AC97C_CHANNEL_B 0x2 #endif /* __SOUND_ATMEL_AC97C_H */ -- cgit v1.2.3 From d54bb9f0c57e39a9a7c8ba523f2c0c1a955d8efb Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:11 +0200 Subject: ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index dd72e00e5ae..21be9c9fbd5 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -312,7 +312,6 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) default: /* TODO: support more than two channels */ return -EINVAL; - break; } ac97c_writel(chip, OCA, word); @@ -374,7 +373,6 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) default: /* TODO: support more than two channels */ return -EINVAL; - break; } ac97c_writel(chip, ICA, word); -- cgit v1.2.3 From 128ed6a9266daac5d7b0e082339742e16caf7caa Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:12 +0200 Subject: ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels This patch will take care not to overwrite OCA and ICA registers when assigning input and output channels. It will also make sure the registers are at a known state when enabling a channel and clean up properly in case of an error. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 21be9c9fbd5..4e8f66d4081 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1,5 +1,5 @@ /* - * Driver for the Atmel AC97C controller + * Driver for Atmel AC97C * * Copyright (C) 2005-2009 Atmel Corporation * @@ -10,6 +10,7 @@ #include #include #include +#include #include #include #include @@ -297,9 +298,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long word = 0; + unsigned long word = ac97c_readl(chip, OCA); int retval; + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + /* assign channels to AC97C channel A */ switch (runtime->channels) { case 1: @@ -323,9 +326,13 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ - default: word &= ~(AC97C_CMR_CEM_LITTLE); break; + default: + word = ac97c_readl(chip, OCA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, OCA, word); + return -EINVAL; } ac97c_writel(chip, CAMR, word); @@ -358,9 +365,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long word = 0; + unsigned long word = ac97c_readl(chip, ICA); int retval; + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + /* assign channels to AC97C channel A */ switch (runtime->channels) { case 1: @@ -384,9 +393,13 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ - default: word &= ~(AC97C_CMR_CEM_LITTLE); break; + default: + word = ac97c_readl(chip, ICA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, ICA, word); + return -EINVAL; } ac97c_writel(chip, CAMR, word); -- cgit v1.2.3 From c42eec0f193ed408118e20d85ea8c2e69c529993 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:13 +0200 Subject: ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter This patch will set a proper maximum bytes for the buffer, which is: channels * bytes per sample * maximum periods * maximum bytes per period. It also sets the minimum periods to 6, a value chosen from testing, with a minimum of 6 periods the system has good time to fill in new audio data without skipping. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 4e8f66d4081..c9bc3458fa2 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -151,10 +151,10 @@ static struct snd_pcm_hardware atmel_ac97c_hw = { .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = 64 * 4096, + .buffer_bytes_max = 2 * 2 * 64 * 2048, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; -- cgit v1.2.3 From df163587eab15a24cc34cf8434a5657416f8a203 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:14 +0200 Subject: ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting This patch will enable interrupts from AC97C and report about error conditions that occurs. On channel A both overrun and underrun will be enabled depending if playback and/or capture are enabled. On the control channel the overrun interrupt is enabled. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 76 insertions(+), 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index c9bc3458fa2..e8484cb9ac6 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -66,6 +66,7 @@ struct atmel_ac97c { /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; + int irq; int opened; int reset_pin; }; @@ -335,8 +336,16 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) return -EINVAL; } + /* Enable underrun interrupt on channel A */ + word |= AC97C_CSR_UNRUN; + ac97c_writel(chip, CAMR, word); + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + /* set variable rate if needed */ if (runtime->rate != 48000) { word = ac97c_readl(chip, MR); @@ -402,8 +411,16 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) return -EINVAL; } + /* Enable overrun interrupt on channel A */ + word |= AC97C_CSR_OVRUN; + ac97c_writel(chip, CAMR, word); + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + /* set variable rate if needed */ if (runtime->rate != 48000) { word = ac97c_readl(chip, MR); @@ -554,6 +571,43 @@ static struct snd_pcm_ops atmel_ac97_capture_ops = { .pointer = atmel_ac97c_capture_pointer, }; +static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) +{ + struct atmel_ac97c *chip = (struct atmel_ac97c *)dev; + irqreturn_t retval = IRQ_NONE; + u32 sr = ac97c_readl(chip, SR); + u32 casr = ac97c_readl(chip, CASR); + u32 cosr = ac97c_readl(chip, COSR); + + if (sr & AC97C_SR_CAEVT) { + dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + casr & AC97C_CSR_OVRUN ? " OVRUN" : "", + casr & AC97C_CSR_RXRDY ? " RXRDY" : "", + casr & AC97C_CSR_UNRUN ? " UNRUN" : "", + casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + casr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !casr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (sr & AC97C_SR_COEVT) { + dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", + cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", + cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + cosr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !cosr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (retval == IRQ_NONE) { + dev_err(&chip->pdev->dev, "spurious interrupt sr 0x%08x " + "casr 0x%08x cosr 0x%08x\n", sr, casr, cosr); + } + + return retval; +} + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; @@ -701,6 +755,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) .read = atmel_ac97c_read, }; int retval; + int irq; regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!regs) { @@ -714,6 +769,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get irq\n"); + return -ENXIO; + } + pclk = clk_get(&pdev->dev, "pclk"); if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); @@ -730,6 +791,13 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip = get_chip(card); + retval = request_irq(irq, atmel_ac97c_interrupt, 0, "AC97C", chip); + if (retval) { + dev_dbg(&pdev->dev, "unable to request irq %d\n", irq); + goto err_request_irq; + } + chip->irq = irq; + spin_lock_init(&chip->lock); strcpy(card->driver, "Atmel AC97C"); @@ -758,6 +826,10 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) snd_card_set_dev(card, &pdev->dev); + /* Enable overrun interrupt from codec channel */ + ac97c_writel(chip, COMR, AC97C_CSR_OVRUN); + ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT); + retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); if (retval) { dev_dbg(&pdev->dev, "could not register on ac97 bus\n"); @@ -820,7 +892,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) retval = snd_card_register(card); if (retval) { dev_dbg(&pdev->dev, "could not register sound card\n"); - goto err_ac97_bus; + goto err_dma; } platform_set_drvdata(pdev, card); @@ -847,6 +919,8 @@ err_ac97_bus: iounmap(chip->regs); err_ioremap: + free_irq(irq, chip); +err_request_irq: snd_card_free(card); err_snd_card_new: clk_disable(pclk); @@ -898,6 +972,7 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) clk_disable(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); + free_irq(chip->irq, chip); if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) dma_release_channel(chip->dma.rx_chan); -- cgit v1.2.3 From 81baf3a7f686c5d22359cb06fc11d20907ba12f8 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:15 +0200 Subject: ALSA: snd-atmel-ac97c: do a proper reset of the external codec This patch will enable the AC97C before resetting the external codec, leaving the AC97C disabled will result in floating I/O lines that can affect the reset procedure. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index e8484cb9ac6..90527c14901 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -730,17 +730,17 @@ static bool filter(struct dma_chan *chan, void *slave) static void atmel_ac97c_reset(struct atmel_ac97c *chip) { - ac97c_writel(chip, MR, AC97C_MR_WRST); + ac97c_writel(chip, MR, 0); + ac97c_writel(chip, MR, AC97C_MR_ENA); + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); if (gpio_is_valid(chip->reset_pin)) { gpio_set_value(chip->reset_pin, 0); /* AC97 v2.2 specifications says minimum 1 us. */ - udelay(10); + udelay(2); gpio_set_value(chip->reset_pin, 1); } - - udelay(1); - ac97c_writel(chip, MR, AC97C_MR_ENA); } static int __devinit atmel_ac97c_probe(struct platform_device *pdev) @@ -826,6 +826,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) snd_card_set_dev(card, &pdev->dev); + atmel_ac97c_reset(chip); + /* Enable overrun interrupt from codec channel */ ac97c_writel(chip, COMR, AC97C_CSR_OVRUN); ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT); @@ -836,8 +838,6 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) goto err_ac97_bus; } - atmel_ac97c_reset(chip); - retval = atmel_ac97c_mixer_new(chip); if (retval) { dev_dbg(&pdev->dev, "could not register ac97 mixer\n"); -- cgit v1.2.3 From bd74a1843e06eef47bdb17452ed363255eb1d6e3 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:16 +0200 Subject: ALSA: snd-atmel-ac97c: cleanup registers when removing driver This patch will set the channel A and control channel mode register to zero before disabling the AC97C peripheral. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 90527c14901..4df9ca40054 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -969,6 +969,10 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) if (gpio_is_valid(chip->reset_pin)) gpio_free(chip->reset_pin); + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); + ac97c_writel(chip, MR, 0); + clk_disable(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); -- cgit v1.2.3 From 23572856e0363a1d4dcf896f59860f86809da7fc Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:17 +0200 Subject: ALSA: snd-atmel-ac97c: replace bus_id with dev_name() This patch replaces the references to bus_id to the new dev_name() API. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/ac97c.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 4df9ca40054..0c0f8771656 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -856,7 +856,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip->dma.rx_chan = dma_request_channel(mask, filter, dws); dev_info(&chip->pdev->dev, "using %s for DMA RX\n", - chip->dma.rx_chan->dev->device.bus_id); + dev_name(&chip->dma.rx_chan->dev->device)); set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); } @@ -872,7 +872,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) chip->dma.tx_chan = dma_request_channel(mask, filter, dws); dev_info(&chip->pdev->dev, "using %s for DMA TX\n", - chip->dma.tx_chan->dev->device.bus_id); + dev_name(&chip->dma.tx_chan->dev->device)); set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); } -- cgit v1.2.3 From 60a56cce7acb6c66fc6c4fdfef0049e73cfdc8be Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 08:21:18 +0200 Subject: ALSA: snd-atmel-abdac: replace bus_id with dev_name() This patch replaces the references to bus_id to the new dev_name() API. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 28b3c7f7cfe..32064fafbc2 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -502,7 +502,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n", - dac->regs, dac->dma.chan->dev->device.bus_id); + dac->regs, dev_name(&dac->dma.chan->dev->device)); return retval; -- cgit v1.2.3 From fa075ed2dc80440bf3e9092d38a66c3227b174c1 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 2 Apr 2009 13:42:26 +0200 Subject: ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4 This patch increases periods_min to 6 from 4, this will remove any hickups where the buffer is not filled fast enough from user space. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 32064fafbc2..f2f41c85422 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -165,7 +165,7 @@ static struct snd_pcm_hardware atmel_abdac_hw = { .buffer_bytes_max = 64 * 4096, .period_bytes_min = 4096, .period_bytes_max = 4096, - .periods_min = 4, + .periods_min = 6, .periods_max = 64, }; -- cgit v1.2.3 From f3cd3f5d341dc5218d0138a67945182e83174af9 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Thu, 2 Apr 2009 19:44:18 +0800 Subject: ALSA: hda - enable SPDIF output for Intel DX58SO board ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both. DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add 0x10 as slave output to the existing 0x06. Reported-by: Jeroen Van Breedam Tested-by: Jeroen Van Breedam Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82097790f6f..f35e58a2d92 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { {} }; +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + static hda_nid_t alc1200_slave_dig_outs[] = { ALC883_DIGOUT_NID, 0, }; @@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), .channel_mode = alc883_3ST_6ch_intel_modes, .need_dac_fix = 1, -- cgit v1.2.3 From 09318c47b6121c8d18cee50ca7e270a8b7dfd274 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 6 Apr 2009 03:50:46 +0200 Subject: ASoC: Fix null dereference in ak4535_remove() ak4535_remove() from sound/soc/codecs/ak4535.c calls i2c_unregister_device() with a possibly null pointer. This bug was found by smatch (http://repo.or.cz/w/smatch.git/). Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 1f63d387a2f..dd338020276 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -659,7 +659,8 @@ static int ak4535_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); + if (codec->control_data) + i2c_unregister_device(codec->control_data); i2c_del_driver(&ak4535_i2c_driver); #endif kfree(codec->private_data); -- cgit v1.2.3 From 5c15a6869a75000fecea61e9985f4753311ec534 Mon Sep 17 00:00:00 2001 From: Anton Vorontsov Date: Sat, 4 Apr 2009 22:33:19 +0400 Subject: ASoC: fsl_dma: Pass the proper device for dma mapping routines The driver should pass a device that specifies internal DMA ops, but substream->pcm is just a logical device, and thus doesn't have arch- specific dma callbacks, therefore following bug appears: Freescale Synchronous Serial Interface (SSI) ASoC Driver ------------[ cut here ]------------ kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237! Oops: Exception in kernel mode, sig: 5 [#1] ... NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108 Call Trace: [df02bde0] [df02be2c] 0xdf02be2c (unreliable) [df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108 [df02be10] [c023a100] fsl_dma_new+0x68/0x124 [df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234 [df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148 [df02bed0] [c023824c] cs4270_probe+0x34/0x124 [df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4 [df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68 [df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80 [df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28 ... This patch fixes the issue by using card's device instead. Signed-off-by: Anton Vorontsov Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 2c4892c853c..b1a3a278819 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -300,7 +300,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, return -ENOMEM; } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { @@ -418,7 +418,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return -EBUSY; } - dma_private = dma_alloc_coherent(substream->pcm->dev, + dma_private = dma_alloc_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), &ld_buf_phys, GFP_KERNEL); if (!dma_private) { dev_err(substream->pcm->card->dev, @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dev_err(substream->pcm->card->dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); return ret; @@ -778,13 +778,13 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) free_irq(dma_private->irq, dma_private); if (dma_private->ld_buf_phys) { - dma_unmap_single(substream->pcm->dev, + dma_unmap_single(substream->pcm->card->dev, dma_private->ld_buf_phys, sizeof(dma_private->link), DMA_TO_DEVICE); } /* Deallocate the fsl_dma_private structure */ - dma_free_coherent(substream->pcm->dev, + dma_free_coherent(substream->pcm->card->dev, sizeof(struct fsl_dma_private), dma_private, dma_private->ld_buf_phys); substream->runtime->private_data = NULL; -- cgit v1.2.3 From 488fe1665fd90f204fbc825b90060b9e0394e4be Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Apr 2009 09:41:40 +0200 Subject: sound: usb-audio: show sample format width in proc file When listing the device's sample formats in the stream? proc file, the sample format number itself is rather obscure, so we better show the format width, too. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c2db0f95968..175c7d1da5c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2147,7 +2147,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %#x\n", fp->format); + snd_iprintf(buffer, " Format: %#x (%d bits)\n", + fp->format, snd_pcm_format_width(fp->format)); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, -- cgit v1.2.3 From 4608eb089b41386e96bd1410326073a6c1c221ba Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Apr 2009 09:42:42 +0200 Subject: sound: usb-audio: remove check_hw_params_convention() This removes the check_hw_params_convention() function because 1) it is not necessary, as the hw_rule_* functions also work correctly (i.e., as no-ops) when the device supports all combinations of the audio format parameters; and 2) it would become too complex when adding a fourth altsetting-dependent hardware parameter, as this would require another three loops to check dependecies with rate/channels/format. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 135 ++++++++------------------------------------------- 1 file changed, 19 insertions(+), 116 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 175c7d1da5c..b7fa0b0b713 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1736,97 +1736,6 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } -#define MAX_MASK 64 - -/* - * check whether the registered audio formats need special hw-constraints - */ -static int check_hw_params_convention(struct snd_usb_substream *subs) -{ - int i; - u32 *channels; - u32 *rates; - u32 cmaster, rmaster; - u32 rate_min = 0, rate_max = 0; - struct list_head *p; - int err = 1; - - channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - if (!channels || !rates) { - err = -ENOMEM; - goto __out; - } - - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - /* unconventional channels? */ - if (f->channels > 32) - goto __out; - /* continuous rate min/max matches? */ - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rate_min && f->rate_min != rate_min) - goto __out; - if (rate_max && f->rate_max != rate_max) - goto __out; - rate_min = f->rate_min; - rate_max = f->rate_max; - } - /* combination of continuous rates and fixed rates? */ - if (rates[f->format] & SNDRV_PCM_RATE_CONTINUOUS) { - if (f->rates != rates[f->format]) - goto __out; - } - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rates[f->format] && rates[f->format] != f->rates) - goto __out; - } - channels[f->format] |= 1 << (f->channels - 1); - rates[f->format] |= f->rates; - /* needs knot? */ - if (f->rates & SNDRV_PCM_RATE_KNOT) - goto __out; - } - /* check whether channels and rates match for all formats */ - cmaster = rmaster = 0; - for (i = 0; i < MAX_MASK; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (rmaster != rates[i] && rmaster && rates[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; - if (rates[i]) - rmaster = rates[i]; - } - /* check whether channels match for all distinct rates */ - memset(channels, 0, MAX_MASK * sizeof(u32)); - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) - continue; - for (i = 0; i < 32; i++) { - if (f->rates & (1 << i)) - channels[i] |= 1 << (f->channels - 1); - } - } - cmaster = 0; - for (i = 0; i < 32; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; - } - err = 0; - - __out: - kfree(channels); - kfree(rates); - return err; -} - /* * If the device supports unusual bit rates, does the request meet these? */ @@ -1909,32 +1818,26 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre 1000, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); - err = check_hw_params_convention(subs); - if (err < 0) + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + -1)) < 0) + return err; + if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) return err; - else if (err) { - hwc_debug("setting extra hw constraints...\n"); - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) - return err; - } return 0; } -- cgit v1.2.3 From 744b89e542b9a16c9afb8a88f623fbe059c88ccb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Apr 2009 09:45:01 +0200 Subject: sound: usb-audio: save data packet interval in audioformat structure The data packet interval needs to be available in the audioformat structure, together with the other audio format parameters, so that it can be used to influence ALSA hardware parameters. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b7fa0b0b713..3f974a64c55 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -121,6 +121,7 @@ struct audioformat { unsigned char attributes; /* corresponding attributes of cs endpoint */ unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ + unsigned char datainterval; /* log_2 of data packet interval */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ @@ -1350,12 +1351,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datapipe = usb_sndisocpipe(dev, ep); else subs->datapipe = usb_rcvisocpipe(dev, ep); - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - subs->datainterval = get_endpoint(alts, 0)->bInterval - 1; - else - subs->datainterval = 0; + subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; subs->fill_max = 0; @@ -2070,6 +2066,9 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + snd_iprintf(buffer, " Data packet interval: %d us\n", + 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } @@ -2563,6 +2562,17 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp return 0; } +static unsigned char parse_datainterval(struct snd_usb_audio *chip, + struct usb_host_interface *alts) +{ + if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && + get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + else + return 0; +} + static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) @@ -2668,6 +2678,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->altset_idx = i; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) @@ -2859,6 +2870,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); @@ -2953,6 +2965,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = 0; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); switch (fp->maxpacksize) { @@ -3020,6 +3033,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]); @@ -3072,6 +3086,7 @@ static int create_ua101_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); -- cgit v1.2.3 From a7d9c0990d5503775784fef7ff44d74d7e3294fd Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Apr 2009 09:48:26 +0200 Subject: sound: usb-audio: allow period sizes less than 1 ms To enable periods shorter than 1 ms, we have to make sure that short periods are only available for alternate settings that have a small enough data packet interval. Furthermore, the code that aligns URBs to USB frames is now superfluous. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 106 +++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 86 insertions(+), 20 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 3f974a64c55..823296d7d57 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -171,7 +171,6 @@ struct snd_usb_substream { unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int fmt_type; /* USB audio format type (1-3) */ - unsigned int packs_per_ms; /* packets per millisecond (for playback) */ unsigned int running: 1; /* running status */ @@ -608,9 +607,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, break; } } - /* finish at the frame boundary at/after the period boundary */ - if (period_elapsed && - (i & (subs->packs_per_ms - 1)) == subs->packs_per_ms - 1) + if (period_elapsed) /* finish at the period boundary */ break; } if (subs->hwptr_done + offs > runtime->buffer_size) { @@ -1068,7 +1065,6 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; - subs->packs_per_ms = packs_per_ms; if (is_playback) { urb_packs = max(nrpacks, 1); @@ -1088,18 +1084,17 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri minsize -= minsize >> 3; minsize = max(minsize, 1u); total_packs = (period_bytes + minsize - 1) / minsize; - /* round up to multiple of packs_per_ms */ - total_packs = (total_packs + packs_per_ms - 1) - & ~(packs_per_ms - 1); /* we need at least two URBs for queueing */ - if (total_packs < 2 * packs_per_ms) { - total_packs = 2 * packs_per_ms; + if (total_packs < 2) { + total_packs = 2; } else { /* and we don't want too long a queue either */ maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); total_packs = min(total_packs, maxpacks); } } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; total_packs = MAX_URBS * urb_packs; } subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; @@ -1564,11 +1559,15 @@ static struct snd_pcm_hardware snd_usb_hardware = #define hwc_debug(fmt, args...) /**/ #endif -static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audioformat *fp) +static int hw_check_valid_format(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *params, + struct audioformat *fp) { struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + unsigned int ptime; /* check the format */ if (!snd_mask_test(fmts, fp->format)) { @@ -1589,6 +1588,14 @@ static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audiof hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min); return 0; } + /* check whether the period time is >= the data packet interval */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + ptime = 125 * (1 << fp->datainterval); + if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { + hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); + return 0; + } + } return 1; } @@ -1607,7 +1614,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->rate_min) @@ -1661,7 +1668,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->channels) @@ -1714,7 +1721,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; fbits |= (1ULL << fp->format); } @@ -1732,6 +1739,44 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } +static int hw_rule_period_time(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + struct audioformat *fp; + struct snd_interval *it; + unsigned char min_datainterval; + unsigned int pmin; + int changed; + + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max); + min_datainterval = 0xff; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + min_datainterval = min(min_datainterval, fp->datainterval); + } + if (min_datainterval == 0xff) { + hwc_debug(" --> get emtpy\n"); + it->empty = 1; + return -EINVAL; + } + pmin = 125 * (1 << min_datainterval); + changed = 0; + if (it->min < pmin) { + it->min = pmin; + it->openmin = 0; + changed = 1; + } + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed); + return changed; +} + /* * If the device supports unusual bit rates, does the request meet these? */ @@ -1777,6 +1822,8 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { struct list_head *p; + unsigned int pt, ptmin; + int param_period_time_if_needed; int err; runtime->hw.formats = subs->formats; @@ -1786,6 +1833,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = 256; runtime->hw.channels_max = 0; runtime->hw.rates = 0; + ptmin = UINT_MAX; /* check min/max rates and channels */ list_for_each(p, &subs->fmt_list) { struct audioformat *fp; @@ -1804,34 +1852,52 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; } + pt = 125 * (1 << fp->datainterval); + ptmin = min(ptmin, pt); } - /* set the period time minimum 1ms */ - /* FIXME: high-speed mode allows 125us minimum period, but many parts - * in the current code assume the 1ms period. - */ + param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; + if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + /* full speed devices have fixed data packet interval */ + ptmin = 1000; + if (ptmin == 1000) + /* if period time doesn't go below 1 ms, no rules needed */ + param_period_time_if_needed = -1; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000, - /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); + ptmin, UINT_MAX); if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, hw_rule_rate, subs, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, -1)) < 0) return err; if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, -1)) < 0) return err; if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, -1)) < 0) return err; + if (param_period_time_if_needed >= 0) { + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + hw_rule_period_time, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, + -1); + if (err < 0) + return err; + } if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) return err; return 0; -- cgit v1.2.3 From bca68467b59a24396554d8dd5979ee363c174854 Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Mon, 6 Apr 2009 18:42:42 +0900 Subject: ALSA: hda - add missing comma in ad1884_slave_vols Signed-off-by: Akinobu Mita Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5bb48ee8b6c..38ad3f7b040 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3256,7 +3256,7 @@ static const char *ad1884_slave_vols[] = { "Mic Playback Volume", "CD Playback Volume", "Internal Mic Playback Volume", - "Docking Mic Playback Volume" + "Docking Mic Playback Volume", /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL -- cgit v1.2.3 From d2e8e52976b9d0a34db529b06952d5187b78af8c Mon Sep 17 00:00:00 2001 From: Deepika Makhija Date: Sat, 4 Apr 2009 18:08:28 +0530 Subject: ALSA: oss - volume control for CSWITCH and CROUTE Added an else part to check SNDRV_MIXER_OSS_PRESENT_CVOLUME for MIC (slot 7) in commit 36c7b833e5d2501142a371e4e75281d3a29fbd6b Similarly, checks and volume control is required for SNDRV_MIXER_OSS_PRESENT_CSWITCH and SNDRV_MIXER_OSS_PRESENT_CROUTE as well. Signed-off-by: Deepika Makhija Signed-off-by: Viral Mehta Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index e570649184e..5dcd8a52697 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -703,19 +703,27 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, if (left || right) { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } else { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } -- cgit v1.2.3 From fd60cc897a6a5093acd9d6554013e679fcc6c5a1 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 6 Apr 2009 09:30:46 -0400 Subject: ALSA: hda - Add VREF powerdown sequence for another board Add powerdown sequence for VREF using a shared jack when the headphone is present and the microphone isn't on. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b34d78b88a8..61996a2f45d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4413,6 +4413,24 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); stac92xx_report_jack(codec, event->nid); + + switch (codec->subsystem_id) { + case 0x103c308f: + if (event->nid == 0xb) { + int pin = AC_PINCTL_IN_EN; + + if (get_pin_presence(codec, 0xa) + && get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + if (!get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + + /* toggle VREF state based on mic + hp pin + * status + */ + stac92xx_auto_set_pinctl(codec, 0x0a, pin); + } + } break; case STAC_VREF_EVENT: data = snd_hda_codec_read(codec, codec->afg, 0, -- cgit v1.2.3