diff options
Diffstat (limited to 'sound')
48 files changed, 236 insertions, 239 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index b37b702a3a6a..5119fdabcb98 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -1110,18 +1110,7 @@ static struct amba_driver aaci_driver = { .id_table = aaci_ids, }; -static int __init aaci_init(void) -{ - return amba_driver_register(&aaci_driver); -} - -static void __exit aaci_exit(void) -{ - amba_driver_unregister(&aaci_driver); -} - -module_init(aaci_init); -module_exit(aaci_exit); +module_amba_driver(aaci_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ARM PrimeCell PL041 Advanced Audio CODEC Interface driver"); diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index d1aa4218f129..48d7c0aa5073 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -17,11 +17,12 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/module.h> +#include <linux/io.h> #include <sound/ac97_codec.h> #include <sound/pxa2xx-lib.h> -#include <asm/irq.h> +#include <mach/irqs.h> #include <mach/regs-ac97.h> #include <mach/audio.h> diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 3a39626a82d6..afef72c4f0d3 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/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/atmel/abdac.c b/sound/atmel/abdac.c index 4fa1dbd8ee83..f7c2bb08055d 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -16,6 +16,7 @@ #include <linux/interrupt.h> #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/types.h> #include <linux/io.h> #include <sound/core.h> @@ -467,15 +468,24 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev) snd_card_set_dev(card, &pdev->dev); if (pdata->dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->dws; dma_cap_mask_t mask; - dws->tx_reg = regs->start + DAC_DATA; - dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - dac->dma.chan = dma_request_channel(mask, filter, dws); + dac->dma.chan = dma_request_channel(mask, filter, &pdata->dws); + if (dac->dma.chan) { + struct dma_slave_config dma_conf = { + .dst_addr = regs->start + DAC_DATA, + .dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .src_maxburst = 1, + .dst_maxburst = 1, + .direction = DMA_MEM_TO_DEV, + .device_fc = false, + }; + + dmaengine_slave_config(dac->dma.chan, &dma_conf); + } } if (!pdata->dws.dma_dev || !dac->dma.chan) { dev_dbg(&pdev->dev, "DMA not available\n"); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 61dade698358..115313ef54d6 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -20,6 +20,7 @@ #include <linux/platform_device.h> #include <linux/mutex.h> #include <linux/gpio.h> +#include <linux/types.h> #include <linux/io.h> #include <sound/core.h> @@ -1014,16 +1015,28 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) if (cpu_is_at32ap7000()) { if (pdata->rx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->rx_dws; dma_cap_mask_t mask; - dws->rx_reg = regs->start + AC97C_CARHR + 2; - dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); chip->dma.rx_chan = dma_request_channel(mask, filter, - dws); + &pdata->rx_dws); + if (chip->dma.rx_chan) { + struct dma_slave_config dma_conf = { + .src_addr = regs->start + AC97C_CARHR + + 2, + .src_addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES, + .src_maxburst = 1, + .dst_maxburst = 1, + .direction = DMA_DEV_TO_MEM, + .device_fc = false, + }; + + dmaengine_slave_config(chip->dma.rx_chan, + &dma_conf); + } dev_info(&chip->pdev->dev, "using %s for DMA RX\n", dev_name(&chip->dma.rx_chan->dev->device)); @@ -1031,16 +1044,28 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) } if (pdata->tx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->tx_dws; dma_cap_mask_t mask; - dws->tx_reg = regs->start + AC97C_CATHR + 2; - dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); chip->dma.tx_chan = dma_request_channel(mask, filter, - dws); + &pdata->tx_dws); + if (chip->dma.tx_chan) { + struct dma_slave_config dma_conf = { + .dst_addr = regs->start + AC97C_CATHR + + 2, + .dst_addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES, + .src_maxburst = 1, + .dst_maxburst = 1, + .direction = DMA_MEM_TO_DEV, + .device_fc = false, + }; + + dmaengine_slave_config(chip->dma.tx_chan, + &dma_conf); + } dev_info(&chip->pdev->dev, "using %s for DMA TX\n", dev_name(&chip->dma.tx_chan->dev->device)); diff --git a/sound/core/init.c b/sound/core/init.c index 068cf08d3ffb..d8ec849af128 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -22,6 +22,7 @@ #include <linux/init.h> #include <linux/sched.h> #include <linux/module.h> +#include <linux/device.h> #include <linux/file.h> #include <linux/slab.h> #include <linux/time.h> diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6e4bfcc14254..1a3070b4e5b5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -24,6 +24,7 @@ #include <linux/module.h> #include <linux/time.h> #include <linux/mutex.h> +#include <linux/device.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 9d8379aedf40..712110561082 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -21,6 +21,7 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/device.h> #include <sound/core.h> #include <sound/initval.h> diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index bbe32d2177d9..dbc550716790 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -46,7 +46,7 @@ The number of ports to be created can be specified via the module parameter "ports". For example, to create four ports, add the - following option in /etc/modprobe.conf: + following option in a configuration file under /etc/modprobe.d/: option snd-seq-dummy ports=4 diff --git a/sound/core/timer.c b/sound/core/timer.c index 8e7561dfc5fc..6ddcf06f52f9 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -24,6 +24,7 @@ #include <linux/slab.h> #include <linux/time.h> #include <linux/mutex.h> +#include <linux/device.h> #include <linux/module.h> #include <linux/string.h> #include <sound/core.h> diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index c8961165277c..fe5ae09ffccb 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -50,7 +50,8 @@ config SND_PCSP before the other sound driver of yours, making the pc-speaker a default sound device. Which is likely not what you want. To make this driver play nicely with other - sound driver, you can add this into your /etc/modprobe.conf: + sound driver, you can add this in a configuration file under + /etc/modprobe.d/ directory: options snd-pcsp index=2 You don't need this driver if you only want your pc-speaker to beep. diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index babaedd242f7..d7ccf28bd66a 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -65,7 +65,7 @@ static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ //static bool enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp = 1; /* Enable ISA PnP detection */ +static bool isapnp = true; /* Enable ISA PnP detection */ #endif static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */ static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */ diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b4a6aa960f4b..8490f59709bb 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } /* diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index eba734560f6f..536c4c0514d3 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate) static int upload_dsp_code(void) { + int ret = 0; + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); #ifndef HAVE_DSPCODEH INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); @@ -1312,7 +1314,8 @@ static int upload_dsp_code(void) memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); - return -ENODEV; + ret = -ENODEV; + goto out; } #ifdef HAVE_DSPCODEH printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); @@ -1320,12 +1323,13 @@ static int upload_dsp_code(void) printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); #endif +out: #ifndef HAVE_DSPCODEH vfree(INITCODE); vfree(PERMCODE); #endif - return 0; + return ret; } #ifdef MSND_CLASSIC @@ -1631,7 +1635,7 @@ static int ide_irq __initdata = 0; static int joystick_io __initdata = 0; /* If we have the digital daugherboard... */ -static int digital __initdata = 0; +static bool digital __initdata = false; #endif static int fifosize __initdata = DEFFIFOSIZE; diff --git a/sound/oss/os.h b/sound/oss/os.h index a1a962d7f67d..75ad0cd0c0ab 100644 --- a/sound/oss/os.h +++ b/sound/oss/os.h @@ -16,7 +16,6 @@ #include <linux/slab.h> #include <linux/ioport.h> #include <asm/page.h> -#include <asm/system.h> #include <linux/vmalloc.h> #include <asm/uaccess.h> #include <linux/poll.h> diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 12ba28e7b933..92ca5bee1860 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -28,7 +28,6 @@ #include <asm/io.h> #include <asm/hardware/iomd.h> #include <asm/irq.h> -#include <asm/system.h> #include "sound_config.h" #include "vidc.h" diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 52468742d9f2..24c430f721d4 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -42,7 +42,6 @@ #include <linux/spinlock.h> #include <linux/bitops.h> -#include <asm/system.h> #include "sound_config.h" #include "waveartist.h" diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 88168044375f..5ca0939e4223 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -2,8 +2,8 @@ config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO menuconfig SND_PCI bool "PCI sound devices" diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 4cc315daeda0..bc86cb726d79 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 2d7d1c2e1d0d..5ef4fe964366 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -39,9 +39,9 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) } -/** Allocated an area of locked memory for bus master DMA operations. +/** Allocate an area of locked memory for bus master DMA operations. -On error, return -ENOMEM, and *pMemArea.size = 0 +If allocation fails, return 1, and *pMemArea.size = 0 */ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) @@ -62,7 +62,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, HPI_DEBUG_LOG(WARNING, "failed to allocate %d bytes locked memory\n", size); p_mem_area->size = 0; - return -ENOMEM; + return 1; } } diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index c5cef113c209..d3fbd0d76c37 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -30,7 +30,6 @@ HPI Operating System Specific macros for Linux Kernel driver #define HPI_BUILD_KERNEL_MODE #include <linux/io.h> -#include <asm/system.h> #include <linux/ioctl.h> #include <linux/kernel.h> #include <linux/string.h> diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index 8afd8b5d1ac7..4439636971eb 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -27,7 +27,6 @@ #include <linux/pci.h> #include <linux/interrupt.h> #include <linux/delay.h> -#include <asm/system.h> #include <asm/io.h> #include <sound/core.h> #include <sound/initval.h> diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9a9f372e1be4..56b4f74c0b13 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -851,6 +851,9 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int single_adc_amp:1; /* adc in-amp takes no index + * (e.g. CX20549 codec) + */ unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b58b4b1687fa..4c054f4486b9 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 254ab5204603..e59e2f059b6e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -651,9 +651,16 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-In caps: "); print_amp_caps(buffer, codec, nid, HDA_INPUT); snd_iprintf(buffer, " Amp-In vals: "); - print_amp_vals(buffer, codec, nid, HDA_INPUT, - wid_caps & AC_WCAP_STEREO, - wid_type == AC_WID_PIN ? 1 : conn_len); + if (wid_type == AC_WID_PIN || + (codec->single_adc_amp && + wid_type == AC_WID_AUD_IN)) + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + 1); + else + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); } if (wid_caps & AC_WCAP_OUT_AMP) { snd_iprintf(buffer, " Amp-Out caps: "); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c6523bbc797..a36488d94aaa 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -141,7 +141,6 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; unsigned int pin_eapd_ctrls:1; - unsigned int single_adc_amp:1; unsigned int adc_switching:1; @@ -687,27 +686,26 @@ static const struct hda_channel_mode cxt5045_modes[1] = { static const struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, } }; static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 5, + .num_items = 4, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, - { "LineIn", 0x3 }, - { "CD", 0x4 }, - { "Mixer", 0x0 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, + { "Line", 0x3 }, + { "Mixer", 0x0 }, } }; static const struct hda_input_mux cxt5045_capture_source_hp530 = { .num_items = 2, .items = { - { "ExtMic", 0x1 }, - { "IntMic", 0x2 }, + { "Mic", 0x1 }, + { "Internal Mic", 0x2 }, } }; @@ -798,10 +796,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), @@ -822,27 +818,15 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { }; static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), {} }; static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), @@ -946,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Output controls */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), /* Modes for retasking pin widgets */ CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), @@ -960,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Loopback mixer controls */ - HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -978,16 +962,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { .put = conexant_mux_enum_put, }, /* Audio input controls */ - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), { } /* end */ }; @@ -1009,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Start with output sum widgets muted and their output gains at min */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Unmute retasking pin widget output buffers since the default * state appears to be output. As the pin mode is changed by the * user the pin mode control will take care of enabling the pin's @@ -1027,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { /* Set ADC connection select to match default mixer setting (mic1 * pin) */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ @@ -1110,7 +1082,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; + codec->single_adc_amp = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -4220,7 +4192,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; - if (spec->single_adc_amp) + if (codec->single_adc_amp) idx = 0; return cx_auto_add_volume_idx(codec, label, pfx, cidx, adc_nid, HDA_INPUT, idx); @@ -4275,7 +4247,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) if (cidx < 0) continue; input_conn[i] = spec->imux_info[i].adc; - if (!spec->single_adc_amp) + if (!codec->single_adc_amp) input_conn[i] |= cidx << 8; if (i > 0 && input_conn[i] != input_conn[0]) multi_connection = 1; @@ -4466,15 +4438,17 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; switch (codec->vendor_id) { case 0x14f15045: - spec->single_adc_amp = 1; + codec->single_adc_amp = 1; break; case 0x14f15051: add_cx5051_fake_mutes(codec); + codec->pin_amp_workaround = 1; break; + default: + codec->pin_amp_workaround = 1; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540cd13f7f15..83f345f3c961 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) struct hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pin_nid; - int pd = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; struct hda_jack_tbl *jack; @@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) pin_nid = jack->nid; jack->jack_dirty = 1; - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pd, eldv); + codec->addr, pin_nid, + !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx < 0) @@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) if (eld->monitor_present) eld_valid = !!(present & AC_PINSENSE_ELDV); - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld_valid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ea2fd654327..2508f8109f11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2717,9 +2717,6 @@ static int alc_auto_fill_adc_caps(struct hda_codec *codec) int max_nums = ARRAY_SIZE(spec->private_adc_nids); int i, nums = 0; - if (spec->shared_mic_hp) - max_nums = 1; /* no multi streams with the shared HP/mic */ - nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { hda_nid_t src; @@ -3401,8 +3398,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (;;) { badness = fill_and_eval_dacs(codec, fill_hardwired, fill_mio_first); - if (badness < 0) + if (badness < 0) { + kfree(best_cfg); return badness; + } debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", cfg->line_out_type, fill_hardwired, fill_mio_first, badness); @@ -3437,7 +3436,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; fill_hardwired = true; continue; - } + } if (cfg->hp_outs > 0 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { cfg->speaker_outs = cfg->line_outs; @@ -3451,7 +3450,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_HP_OUT; fill_hardwired = true; continue; - } + } break; } @@ -4076,6 +4075,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (spec->dyn_adc_switch) return; + again: nums = 0; for (n = 0; n < spec->num_adc_nids; n++) { hda_nid_t cap = spec->private_capsrc_nids[n]; @@ -4096,6 +4096,11 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (!nums) { /* check whether ADC-switch is possible */ if (!alc_check_dyn_adc_switch(codec)) { + if (spec->shared_mic_hp) { + spec->shared_mic_hp = 0; + spec->private_imux[0].num_items = 1; + goto again; + } printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" " using fallback 0x%x\n", codec->chip_name, spec->private_adc_nids[0]); @@ -4113,7 +4118,7 @@ static void alc_remove_invalid_adc_nids(struct hda_codec *codec) if (spec->auto_mic) alc_auto_mic_check_imux(codec); /* check auto-mic setups */ - else if (spec->input_mux->num_items == 1) + else if (spec->input_mux->num_items == 1 || spec->shared_mic_hp) spec->num_adc_nids = 1; /* reduce to a single ADC */ } @@ -4420,7 +4425,7 @@ static int alc_parse_auto_config(struct hda_codec *codec, static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; + static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } @@ -5266,7 +5271,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x16, 0x99130111 }, /* CLFE speaker */ { 0x17, 0x99130112 }, /* surround speaker */ { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC882_FIXUP_ACER_ASPIRE_8930G] = { .type = ALC_FIXUP_PINS, @@ -5309,7 +5316,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC885_FIXUP_MACPRO_GPIO] = { .type = ALC_FIXUP_FUNC, @@ -5356,6 +5365,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), + SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), @@ -5381,6 +5391,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), @@ -5396,6 +5407,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, + {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {} +}; + /* * BIOS auto configuration */ @@ -5436,7 +5454,8 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, + alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -6076,7 +6095,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), @@ -6293,7 +6312,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec, { if (action == ALC_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; -} +} static const struct alc_fixup alc861_fixups[] = { [ALC861_FIXUP_FSC_AMILO_PI1505] = { @@ -6711,7 +6730,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f8e10ced244a..b3e24f289421 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 d1926266fe00..8e92fb88ed09 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 fe7fbaeb7146..7c49642af052 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3629,7 +3629,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case 2: case 3: wm8994->hubs.dcs_codes_l = -9; - wm8994->hubs.dcs_codes_r = -5; + wm8994->hubs.dcs_codes_r = -7; break; default: break; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index afbabf427f27..3fea5a15ffe8 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) { struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts_86xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -142,9 +142,9 @@ static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) { struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts_86xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_86xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 46623405a2ce..982a1c944983 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -46,7 +46,7 @@ * ch: The channel on the DMA controller (0, 1, 2, or 3) * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx) */ -static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts, +static inline void guts_set_dmuxcr(struct ccsr_guts __iomem *guts, unsigned int co, unsigned int ch, unsigned int device) { unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); @@ -90,9 +90,9 @@ static int p1022_ds_machine_probe(struct snd_soc_card *card) { struct machine_data *mdata = container_of(card, struct machine_data, card); - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; @@ -164,9 +164,9 @@ static int p1022_ds_machine_remove(struct snd_soc_card *card) { struct machine_data *mdata = container_of(card, struct machine_data, card); - struct ccsr_guts_85xx __iomem *guts; + struct ccsr_guts __iomem *guts; - guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); if (!guts) { dev_err(card->dev, "could not map global utilities\n"); return -ENOMEM; diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index a839494c5ea8..f23700359c67 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -40,12 +40,6 @@ static void __iomem *audmux_base; #ifdef CONFIG_DEBUG_FS static struct dentry *audmux_debugfs_root; -static int audmux_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - /* There is an annoying discontinuity in the SSI numbering with regard * to the Linux number of the devices */ static const char *audmux_port_string(int port) @@ -79,14 +73,17 @@ 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); @@ -142,7 +139,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, } static const struct file_operations audmux_debugfs_fops = { - .open = audmux_open_file, + .open = simple_open, .read = audmux_read_file, .llseek = default_llseek, }; @@ -158,7 +155,7 @@ 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 +234,13 @@ 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 e43c8fa2788b..6b818de2fc03 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 6ca1f46d84a4..e373fbbc97a0 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 12be05b16880..53f4fd8feced 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 49fe63ce51f7..7d4fa8ed6699 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 4800d5fe568d..06ea2744cc88 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 609abd51e55f..d08583790d23 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 f3417f2311b8..fe3995ce9b38 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 0193e595d415..5cfcc655e95f 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 a4deebc0801a..accdcb7d4d9d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -201,12 +201,6 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); #ifdef CONFIG_DEBUG_FS -static int codec_reg_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { @@ -264,7 +258,7 @@ static ssize_t codec_reg_write_file(struct file *file, } static const struct file_operations codec_reg_fops = { - .open = codec_reg_open_file, + .open = simple_open, .read = codec_reg_read_file, .write = codec_reg_write_file, .llseek = default_llseek, @@ -1087,6 +1081,8 @@ 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/soc-dapm.c b/sound/soc/soc-dapm.c index 6241490fff30..5cbd2d7623b8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1544,12 +1544,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } #ifdef CONFIG_DEBUG_FS -static int dapm_widget_power_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - static ssize_t dapm_widget_power_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -1613,17 +1607,11 @@ static ssize_t dapm_widget_power_read_file(struct file *file, } static const struct file_operations dapm_widget_power_fops = { - .open = dapm_widget_power_open_file, + .open = simple_open, .read = dapm_widget_power_read_file, .llseek = default_llseek, }; -static int dapm_bias_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - static ssize_t dapm_bias_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { @@ -1654,7 +1642,7 @@ static ssize_t dapm_bias_read_file(struct file *file, char __user *user_buf, } static const struct file_operations dapm_bias_fops = { - .open = dapm_bias_open_file, + .open = simple_open, .read = dapm_bias_read_file, .llseek = default_llseek, }; diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4420b7030c83..475695234b3d 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -143,7 +143,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) direction = snd_pcm_substream_to_dma_direction(substream); prtd->pos = 0; - desc = chan->device->device_prep_dma_cyclic(chan, + desc = dmaengine_prep_dma_cyclic(chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream), direction); diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de52540..e53349912b2e 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 475428cf270e..9ff2c601445f 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 21554611557c..b609d2c64c55 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); |