diff options
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/ak4642.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/sgtl5000.c | 25 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 25 | ||||
-rw-r--r-- | sound/soc/imx/imx-audmux.c | 13 | ||||
-rw-r--r-- | sound/soc/imx/imx-pcm-dma-mx2.c | 3 | ||||
-rw-r--r-- | sound/soc/mxs/mxs-pcm.c | 2 | ||||
-rw-r--r-- | sound/soc/mxs/mxs-saif.c | 2 | ||||
-rw-r--r-- | sound/soc/omap/ams-delta.c | 34 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-ac97.c | 1 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.c | 1 | ||||
-rw-r--r-- | sound/soc/samsung/Kconfig | 12 | ||||
-rw-r--r-- | sound/soc/sh/siu_pcm.c | 4 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 2 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_i2s.c | 6 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_spdif.c | 4 | ||||
-rw-r--r-- | sound/soc/txx9/txx9aclc.c | 2 |
16 files changed, 51 insertions, 87 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f8e10ce..b3e24f2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d192626..8e92fb8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), }; @@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 10d2789..7c49642 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2181,26 +2181,9 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { switch (control->type) { - case WM8994: - if (wm8994->revision < 4) { - /* Tweak DC servo and DSP - * configuration for improved - * performance. */ - snd_soc_write(codec, 0x102, 0x3); - snd_soc_write(codec, 0x56, 0x3); - snd_soc_write(codec, 0x817, 0); - snd_soc_write(codec, 0x102, 0); - } - break; - case WM8958: if (wm8994->revision == 0) { /* Optimise performance for rev A */ - snd_soc_write(codec, 0x102, 0x3); - snd_soc_write(codec, 0xcb, 0x81); - snd_soc_write(codec, 0x817, 0); - snd_soc_write(codec, 0x102, 0); - snd_soc_update_bits(codec, WM8958_CHARGE_PUMP_2, WM8958_CP_DISCH, @@ -2208,13 +2191,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, } break; - case WM1811: - if (wm8994->revision < 2) { - snd_soc_write(codec, 0x102, 0x3); - snd_soc_write(codec, 0x5d, 0x7e); - snd_soc_write(codec, 0x5e, 0x0); - snd_soc_write(codec, 0x102, 0x0); - } + default: break; } diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index a839494..0fe66c3 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -79,14 +79,17 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + if (!audmux_base) + return -ENOSYS; + if (audmux_clk) - clk_enable(audmux_clk); + clk_prepare_enable(audmux_clk); ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); if (audmux_clk) - clk_disable(audmux_clk); + clk_disable_unprepare(audmux_clk); ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); @@ -158,7 +161,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) @@ -237,13 +240,13 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, return -ENOSYS; if (audmux_clk) - clk_enable(audmux_clk); + clk_prepare_enable(audmux_clk); writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); if (audmux_clk) - clk_disable(audmux_clk); + clk_disable_unprepare(audmux_clk); return 0; } diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index e43c8fa..6b818de 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dmaengine.h> +#include <linux/types.h> #include <sound/core.h> #include <sound/initval.h> @@ -58,6 +59,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; + slave_config.device_fc = false; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_maxburst = dma_params->burstsize; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 6ca1f46..e373fbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -28,6 +28,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/dmaengine.h> +#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/initval.h> @@ -36,7 +37,6 @@ #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include <mach/dma.h> #include "mxs-pcm.h" struct mxs_pcm_dma_data { diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 12be05b..53f4fd8 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -24,12 +24,12 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/time.h> +#include <linux/fsl/mxs-dma.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/saif.h> -#include <mach/dma.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 49fe63c..7d4fa8e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = { }; -/* Board specific codec bias level control */ -static int ams_delta_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - case SND_SOC_BIAS_STANDBY: - if (card->dapm.bias_level == SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - AMS_DELTA_LATCH2_MODEM_NRESET); - break; - case SND_SOC_BIAS_OFF: - if (card->dapm.bias_level != SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - 0); - } - card->dapm.bias_level = level; - - return 0; -} - /* Digital mute implemented using modem/CPU multiplexer. * Shares hardware with codec config pulse generation */ static bool ams_delta_muted = 1; @@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) ams_delta_ops.shutdown = ams_delta_shutdown; } - /* Set codec bias level */ - ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); - /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ ret = snd_soc_jack_new(rtd->codec, "hook_switch", @@ -598,7 +572,6 @@ static struct snd_soc_card ams_delta_audio_card = { .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, - .set_bias_level = ams_delta_set_bias_level, }; /* Module init/exit */ @@ -635,7 +608,7 @@ err: platform_device_put(ams_delta_audio_platform_device); return ret; } -module_init(ams_delta_module_init); +late_initcall(ams_delta_module_init); static void __exit ams_delta_module_exit(void) { @@ -647,11 +620,6 @@ static void __exit ams_delta_module_exit(void) ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); - /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, - &ams_delta_audio_card.rtd[0].codec->dapm, - SND_SOC_BIAS_STANDBY); - platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 4800d5f..06ea274 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -11,6 +11,7 @@ */ #include <linux/init.h> +#include <linux/io.h> #include <linux/module.h> #include <linux/platform_device.h> diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd5..d085837 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f3417f2..fe3995c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,8 +1,8 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 + depends on ARCH_S3C24XX || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210 || ARCH_S5P64X0 || ARCH_EXYNOS4 select S3C64XX_DMA if ARCH_S3C64XX - select S3C2410_DMA if ARCH_S3C2410 + select S3C2410_DMA if ARCH_S3C24XX help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -84,7 +84,7 @@ config SND_SOC_SAMSUNG_SMDK2443_WM9710 config SND_SOC_SAMSUNG_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_SOC_SAMSUNG && ARCH_S3C2410 + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_CODEC @@ -95,7 +95,7 @@ config SND_SOC_SAMSUNG_LN2440SBC_ALC650 config SND_SOC_SAMSUNG_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_SOC_SAMSUNG && ARCH_S3C2410 + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX select SND_S3C24XX_I2S select SND_SOC_L3 select SND_SOC_UDA134X @@ -107,14 +107,14 @@ config SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on SND_SOC_SAMSUNG && ARCH_S3C2410 + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX select SND_S3C24XX_I2S select SND_SOC_TLV320AIC23 select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_HERMES tristate "SoC I2S Audio support for Simtec Hermes board" - depends on SND_SOC_SAMSUNG && ARCH_S3C2410 + depends on SND_SOC_SAMSUNG && ARCH_S3C24XX select SND_S3C24XX_I2S select SND_SOC_TLV320AIC3X select SND_SOC_SAMSUNG_SIMTEC diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index 0193e59..5cfcc65 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -130,7 +130,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info, sg_dma_len(&sg) = size; sg_dma_address(&sg) = buff; - desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + desc = dmaengine_prep_slave_sg(siu_stream->chan, &sg, 1, DMA_MEM_TO_DEV, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dev, "Failed to allocate a dma descriptor\n"); @@ -180,7 +180,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info, sg_dma_len(&sg) = size; sg_dma_address(&sg) = buff; - desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan, + desc = dmaengine_prep_slave_sg(siu_stream->chan, &sg, 1, DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dev, "Failed to allocate dma descriptor\n"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc..8d2ebf5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1087,6 +1087,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de..e533499 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused) struct tegra_i2s *i2s = s->private; int i; + clk_enable(i2s->clk_i2s); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_i2s_read(i2s, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(i2s->clk_i2s); + return 0; } @@ -112,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) { } diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 475428c..9ff2c60 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused) struct tegra_spdif *spdif = s->private; int i; + clk_enable(spdif->clk_spdif_out); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_spdif_read(spdif, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(spdif->clk_spdif_out); + return 0; } diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 2155461..b609d2c 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -132,7 +132,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)), dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1)); sg_dma_address(&sg) = buf_dma_addr; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); |