diff options
author | 2012-04-11 11:07:38 -0700 | |
---|---|---|
committer | 2012-04-11 11:07:38 -0700 | |
commit | a1ada086062101533eb0f841d3884137688091ec (patch) | |
tree | 3dd45239db0eaaf7693e5bae75f0c8b61466bb6e | |
parent | Merge branch 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git... (diff) | |
parent | ALSA: hda - hide HDMI/ELD printks unless snd.debug=2 (diff) | |
download | linux-a1ada086062101533eb0f841d3884137688091ec.tar.gz linux-a1ada086062101533eb0f841d3884137688091ec.tar.bz2 linux-a1ada086062101533eb0f841d3884137688091ec.zip |
Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
- A series of fixes for Conexant 20549 HD-audio codec chip
- A workaround for HDMI hotplug debug prints that annoyed people
- A fix for the new support of platform DAPM contexts
- Many driver-specific minor fixes
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
ALSA: sound/isa/sscape.c: add missing resource-release code
sound: sound/oss/msnd_pinnacle.c: add vfrees
ALSA: hda - clean up CX20549 test mixer setup
ALSA: hda - CX20549 doesn't need pin_amp_workaround.
ALSA: hda - Remove CD control from model=benq for CX20549
ALSA: hda - fix record volume controls of CX20459 ("Venice")
ALSA: hda - Rename capture sources of CX20549 to match common conventions
ALSA: hda - Fix proc output for ADC amp values of CX20549
ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
ASoC: set idle_bias_off=1 for all platform DAPM contexts
ASoC: imx-audmux: Check for NULL pointer
ASoC: imx-audmux: Fix ssi port numbers in sysfs
ASoC: ak4642: fixup: mute needs +1 step
MAINTAINERS: Don't list everyone working on Wolfson drivers
MAINTAINERS: Add missing ASoC OMAP co-maintainer
ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
ASoC: tegra: ensure clocks are enabled when touching registers
ASoC: sgtl5000: Enable VAG when DAC/ADC up
ALSA: asihpi - fix return value of hpios_locked_mem_alloc()
-rw-r--r-- | MAINTAINERS | 4 | ||||
-rw-r--r-- | include/sound/core.h | 10 | ||||
-rw-r--r-- | sound/isa/sscape.c | 6 | ||||
-rw-r--r-- | sound/oss/msnd_pinnacle.c | 8 | ||||
-rw-r--r-- | sound/pci/asihpi/hpi_internal.h | 4 | ||||
-rw-r--r-- | sound/pci/asihpi/hpios.c | 10 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 3 | ||||
-rw-r--r-- | sound/pci/hda/hda_eld.c | 6 | ||||
-rw-r--r-- | sound/pci/hda/hda_proc.c | 13 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 108 | ||||
-rw-r--r-- | sound/pci/hda/patch_hdmi.c | 9 | ||||
-rw-r--r-- | sound/soc/codecs/ak4642.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/sgtl5000.c | 25 | ||||
-rw-r--r-- | sound/soc/imx/imx-audmux.c | 5 | ||||
-rw-r--r-- | sound/soc/pxa/pxa2xx-i2s.c | 1 | ||||
-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 |
18 files changed, 120 insertions, 106 deletions
diff --git a/MAINTAINERS b/MAINTAINERS index 2dcfca850639..a1270978eb41 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4803,6 +4803,7 @@ F: arch/arm/mach-omap2/clockdomain2xxx_3xxx.c F: arch/arm/mach-omap2/clockdomain44xx.c OMAP AUDIO SUPPORT +M: Peter Ujfalusi <peter.ujfalusi@ti.com> M: Jarkko Nikula <jarkko.nikula@bitmer.com> L: alsa-devel@alsa-project.org (subscribers-only) L: linux-omap@vger.kernel.org @@ -7461,8 +7462,7 @@ F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS M: Mark Brown <broonie@opensource.wolfsonmicro.com> -M: Ian Lartey <ian@opensource.wolfsonmicro.com> -M: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> +L: patches@opensource.wolfsonmicro.com T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices diff --git a/include/sound/core.h b/include/sound/core.h index b6e0f57d451d..bc056687f647 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -325,6 +325,13 @@ void release_and_free_resource(struct resource *res); /* --- */ +/* sound printk debug levels */ +enum { + SND_PR_ALWAYS, + SND_PR_DEBUG, + SND_PR_VERBOSE, +}; + #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) __printf(4, 5) void __snd_printk(unsigned int level, const char *file, int line, @@ -354,6 +361,8 @@ void __snd_printk(unsigned int level, const char *file, int line, */ #define snd_printd(fmt, args...) \ __snd_printk(1, __FILE__, __LINE__, fmt, ##args) +#define _snd_printd(level, fmt, args...) \ + __snd_printk(level, __FILE__, __LINE__, fmt, ##args) /** * snd_BUG - give a BUG warning message and stack trace @@ -383,6 +392,7 @@ void __snd_printk(unsigned int level, const char *file, int line, #else /* !CONFIG_SND_DEBUG */ #define snd_printd(fmt, args...) do { } while (0) +#define _snd_printd(level, fmt, args...) do { } while (0) #define snd_BUG() do { } while (0) static inline int __snd_bug_on(int cond) { 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 2c79d60a725f..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 diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 8c63200cf339..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 @@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned. If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle. */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, /**< memory handle */ u32 size, /**< Size in bytes to allocate */ struct pci_dev *p_os_reference diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 87f4385fe8c7..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 @@ -39,11 +39,11 @@ 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 */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) { /*?? any benefit in using managed dmam_alloc_coherent? */ @@ -62,7 +62,7 @@ int 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/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/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/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 1765a197acb0..f23700359c67 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -73,6 +73,9 @@ 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_prepare_enable(audmux_clk); @@ -152,7 +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)) 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/soc-core.c b/sound/soc/soc-core.c index e19c24ade414..accdcb7d4d9d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1081,6 +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/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; } |