diff options
Diffstat (limited to 'sound')
58 files changed, 866 insertions, 587 deletions
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index f0ebc971..1dd66dd 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter, client = i2c_new_device(adapter, &info); if (!client) return -ENODEV; + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, and then there + * is no point in keeping the device instantiated. + */ + if (!client->driver) { + i2c_unregister_device(client); + return -ENODEV; + } /* * Let i2c-core delete that device on driver removal. diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index dc78272..1f0f821 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; + writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4e34d19..b4b48af 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -137,9 +137,9 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) return 0; } -static int pxa2xx_ac97_suspend(struct platform_device *dev, pm_message_t state) +static int pxa2xx_ac97_suspend(struct device *dev) { - struct snd_card *card = platform_get_drvdata(dev); + struct snd_card *card = dev_get_drvdata(dev); int ret = 0; if (card) @@ -148,9 +148,9 @@ static int pxa2xx_ac97_suspend(struct platform_device *dev, pm_message_t state) return ret; } -static int pxa2xx_ac97_resume(struct platform_device *dev) +static int pxa2xx_ac97_resume(struct device *dev) { - struct snd_card *card = platform_get_drvdata(dev); + struct snd_card *card = dev_get_drvdata(dev); int ret = 0; if (card) @@ -159,9 +159,10 @@ static int pxa2xx_ac97_resume(struct platform_device *dev) return ret; } -#else -#define pxa2xx_ac97_suspend NULL -#define pxa2xx_ac97_resume NULL +static struct dev_pm_ops pxa2xx_ac97_pm_ops = { + .suspend = pxa2xx_ac97_suspend, + .resume = pxa2xx_ac97_resume, +}; #endif static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) @@ -241,11 +242,12 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_probe, .remove = __devexit_p(pxa2xx_ac97_remove), - .suspend = pxa2xx_ac97_suspend, - .resume = pxa2xx_ac97_resume, .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, +#ifdef CONFIG_PM + .pm = &pxa2xx_ac97_pm_ops, +#endif }, }; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59e5fbe..ab73edf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1387,11 +1387,6 @@ static struct action_ops snd_pcm_action_drain_init = { .post_action = snd_pcm_post_drain_init }; -struct drain_rec { - struct snd_pcm_substream *substream; - wait_queue_t wait; -}; - static int snd_pcm_drop(struct snd_pcm_substream *substream); /* @@ -1407,10 +1402,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_pcm_substream *s; + wait_queue_t wait; int result = 0; - int i, num_drecs; int nonblock = 0; - struct drain_rec *drec, drec_tmp, *d; card = substream->pcm->card; runtime = substream->runtime; @@ -1433,38 +1427,10 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } else if (substream->f_flags & O_NONBLOCK) nonblock = 1; - if (nonblock) - goto lock; /* no need to allocate waitqueues */ - - /* allocate temporary record for drain sync */ down_read(&snd_pcm_link_rwsem); - if (snd_pcm_stream_linked(substream)) { - drec = kmalloc(substream->group->count * sizeof(*drec), GFP_KERNEL); - if (! drec) { - up_read(&snd_pcm_link_rwsem); - snd_power_unlock(card); - return -ENOMEM; - } - } else - drec = &drec_tmp; - - /* count only playback streams */ - num_drecs = 0; - snd_pcm_group_for_each_entry(s, substream) { - runtime = s->runtime; - if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - d = &drec[num_drecs++]; - d->substream = s; - init_waitqueue_entry(&d->wait, current); - add_wait_queue(&runtime->sleep, &d->wait); - } - } - up_read(&snd_pcm_link_rwsem); - - lock: snd_pcm_stream_lock_irq(substream); /* resume pause */ - if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) + if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); /* pre-start/stop - all running streams are changed to DRAINING state */ @@ -1479,25 +1445,35 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, for (;;) { long tout; + struct snd_pcm_runtime *to_check; if (signal_pending(current)) { result = -ERESTARTSYS; break; } - /* all finished? */ - for (i = 0; i < num_drecs; i++) { - runtime = drec[i].substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) + /* find a substream to drain */ + to_check = NULL; + snd_pcm_group_for_each_entry(s, substream) { + if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + runtime = s->runtime; + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + to_check = runtime; break; + } } - if (i == num_drecs) - break; /* yes, all drained */ - + if (!to_check) + break; /* all drained */ + init_waitqueue_entry(&wait, current); + add_wait_queue(&to_check->sleep, &wait); set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); + up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); tout = schedule_timeout(10 * HZ); snd_power_lock(card); + down_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); + remove_wait_queue(&to_check->sleep, &wait); if (tout == 0) { if (substream->runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) result = -ESTRPIPE; @@ -1512,16 +1488,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, unlock: snd_pcm_stream_unlock_irq(substream); - - if (!nonblock) { - for (i = 0; i < num_drecs; i++) { - d = &drec[i]; - runtime = d->substream->runtime; - remove_wait_queue(&runtime->sleep, &d->wait); - } - if (drec != &drec_tmp) - kfree(drec); - } + up_read(&snd_pcm_link_rwsem); snd_power_unlock(card); return result; @@ -3018,7 +2985,7 @@ static int snd_pcm_mmap_status_fault(struct vm_area_struct *area, return 0; } -static struct vm_operations_struct snd_pcm_vm_ops_status = +static const struct vm_operations_struct snd_pcm_vm_ops_status = { .fault = snd_pcm_mmap_status_fault, }; @@ -3057,7 +3024,7 @@ static int snd_pcm_mmap_control_fault(struct vm_area_struct *area, return 0; } -static struct vm_operations_struct snd_pcm_vm_ops_control = +static const struct vm_operations_struct snd_pcm_vm_ops_control = { .fault = snd_pcm_mmap_control_fault, }; @@ -3127,7 +3094,7 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, return 0; } -static struct vm_operations_struct snd_pcm_vm_ops_data = +static const struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, @@ -3151,7 +3118,7 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static struct vm_operations_struct snd_pcm_vm_ops_data_mmio = +static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 6e7d09a..7d722a0 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; extern int use_internal_drums; +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan); /* * The next table looks magical, but it certainly is not. Its values have * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception @@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data) int again = 0; int i; - spin_lock_irqsave(&opl3->sys_timer_lock, flags); + spin_lock_irqsave(&opl3->voice_lock, flags); for (i = 0; i < opl3->max_voices; i++) { struct snd_opl3_voice *vp = &opl3->voices[i]; if (vp->state > 0 && vp->note_off_check) { if (vp->note_off == jiffies) - snd_opl3_note_off(opl3, vp->note, 0, vp->chan); + snd_opl3_note_off_unsafe(opl3, vp->note, 0, + vp->chan); else again++; } } + spin_unlock_irqrestore(&opl3->voice_lock, flags); + + spin_lock_irqsave(&opl3->sys_timer_lock, flags); if (again) { opl3->tlist.expires = jiffies + 1; /* invoke again */ add_timer(&opl3->tlist); @@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* * Release a note in response to a midi note off. */ -void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan) { struct snd_opl3 *opl3; int voice; struct snd_opl3_voice *vp; - unsigned long flags; - opl3 = p; #ifdef DEBUG_MIDI @@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan chan->number, chan->midi_program, note); #endif - spin_lock_irqsave(&opl3->voice_lock, flags); - if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel && use_internal_drums) { snd_opl3_drum_switch(opl3, note, vel, 0, chan); - spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } /* this loop will hopefully kill all extra voices, because @@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan snd_opl3_kill_voice(opl3, voice); } } +} + +void snd_opl3_note_off(void *p, int note, int vel, + struct snd_midi_channel *chan) +{ + struct snd_opl3 *opl3 = p; + unsigned long flags; + + spin_lock_irqsave(&opl3->voice_lock, flags); + snd_opl3_note_off_unsafe(p, note, vel, chan); spin_unlock_irqrestore(&opl3->voice_lock, flags); } diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index c52691c..9a88cdf 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev) return 0; } -static int __exit hal2_remove(struct platform_device *pdev) +static int __devexit hal2_remove(struct platform_device *pdev) { struct snd_card *card = platform_get_drvdata(pdev); diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index e497525..8691f4c 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) return 0; } -static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +static int __devexit snd_sgio2audio_remove(struct platform_device *pdev) { struct snd_card *card = platform_get_drvdata(pdev); diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 1edab7b..3136c88 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -110,9 +110,6 @@ static void start_adc(struct cs4297a_state *s); // rather than 64k as some of the games work more responsively. // log base 2( buff sz = 32k). -//static unsigned long defaultorder = 3; -//MODULE_PARM(defaultorder, "i"); - // // Turn on/off debugging compilation by commenting out "#define CSDEBUG" // diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 1075344..8db6aef 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -100,9 +100,6 @@ def_tmr_open(int dev, int mode) curr_tempo = 60; curr_timebase = 100; opened = 1; - - ; - { def_tmr.expires = (1) + jiffies; add_timer(&def_tmr); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 24585c6..4e2b925 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC), /* Leadtek Winfast tv 2000xp delux */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC), + /* Pinnacle PCTV */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC), /* Voodoo TV 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC), /* Askey Computer Corp. MagicTView'99 */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index b1b3a64..7545464 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state) static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state) { - return atc_daio_unmute(atc, state, LINEO4); + return atc_daio_unmute(atc, state, LINEO2); } static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state) @@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state) static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state) { - return atc_daio_unmute(atc, state, LINEO2); + return atc_daio_unmute(atc, state, LINEO4); } static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index da2065c..1305f7c 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip) Control interface ******************************************************************************/ -#ifndef ECHOCARD_HAS_VMIXER +#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN) /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, @@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, return changed; } +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN +/* On the Mia this one controls the line-out volume */ +static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { + .name = "Line Playback Volume", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = snd_echo_output_gain_info, + .get = snd_echo_output_gain_get, + .put = snd_echo_output_gain_put, + .tlv = {.p = db_scale_output_gain}, +}; +#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; - #endif +#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */ + #ifdef ECHOCARD_HAS_INPUT_GAIN @@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; -#else - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0) +#ifdef ECHOCARD_HAS_LINE_OUT_GAIN + err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_echo_line_output_gain, chip)); + if (err < 0) goto ctl_error; #endif +#else /* ECHOCARD_HAS_VMIXER */ + err = snd_ctl_add(chip->card, + snd_ctl_new1(&snd_echo_pcm_output_gain, chip)); + if (err < 0) + goto ctl_error; +#endif /* ECHOCARD_HAS_VMIXER */ #ifdef ECHOCARD_HAS_INPUT_GAIN if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0) diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index f3b9b45..f05c8c0 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -29,6 +29,7 @@ #define ECHOCARD_HAS_ADAT FALSE #define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 #define ECHOCARD_HAS_MIDI +#define ECHOCARD_HAS_LINE_OUT_GAIN /* Pipe indexes */ #define PX_ANALOG_OUT 0 /* 8 */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 20a66f8..c9ad182 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) * white-list for enable_msi */ static struct snd_pci_quirk msi_white_list[] __devinitdata = { + SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1), SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1), {} }; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 215e72a..2d603f6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) } /* + * HP Touchsmart + * port-A (0x11) - front hp-out + * port-B (0x14) - unused + * port-C (0x15) - unused + * port-D (0x12) - rear line out + * port-E (0x1c) - front mic-in + * port-F (0x16) - Internal speakers + * digital-mic (0x17) - Internal mic + */ + +static struct hda_verb ad1984a_touchsmart_verbs[] = { + /* DACs; unmute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ + /* Port-A (HP) mixer - route only from analog mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-A (HP) pin - always unmuted */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Port-E (int speaker) mixer - route only from analog mixer */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, + /* Port-E pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* Port-F (int speaker) mixer - route only from analog mixer */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-F pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* capture sources */ + /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* unsolicited event for pin-sense */ + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ + /* internal mic - dmic */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* set magic COEFs for dmic */ + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), +/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), + { } /* end */ +}; + +/* switch to external mic if plugged */ +static void ad1984a_touchsmart_automic(struct hda_codec *codec) +{ + if (snd_hda_codec_read(codec, 0x1c, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x4); + } else { + snd_hda_codec_write(codec, 0x0c, 0, + AC_VERB_SET_CONNECT_SEL, 0x5); + } +} + + +/* unsolicited event for HP jack sensing */ +static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_hp_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1984a_touchsmart_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1984a_touchsmart_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_hp_automute(codec); + ad1984a_touchsmart_automic(codec); + return 0; +} + + +/* */ enum { @@ -4039,6 +4160,7 @@ enum { AD1884A_LAPTOP, AD1884A_MOBILE, AD1884A_THINKPAD, + AD1984A_TOUCHSMART, AD1884A_MODELS }; @@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { [AD1884A_LAPTOP] = "laptop", [AD1884A_MOBILE] = "mobile", [AD1884A_THINKPAD] = "thinkpad", + [AD1984A_TOUCHSMART] = "touchsmart", }; static struct snd_pci_quirk ad1884a_cfg_tbl[] = { @@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), + SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), {} }; @@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; codec->patch_ops.init = ad1984a_thinkpad_init; break; + case AD1984A_TOUCHSMART: + spec->mixers[0] = ad1984a_touchsmart_mixers; + spec->init_verbs[0] = ad1984a_touchsmart_verbs; + spec->multiout.dig_out_nid = 0; + codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; + codec->patch_ops.init = ad1984a_touchsmart_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; } return 0; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9d899ed..3fbbc8c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = { }; static struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 3, + .num_items = 5, .items = { { "IntMic", 0x1 }, { "ExtMic", 0x2 }, { "LineIn", 0x3 }, + { "CD", 0x4 }, + { "Mixer", 0x0 }, } }; @@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { }; static 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), + {} }; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index c8435c9..9fb6027 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -29,6 +29,9 @@ #include "hda_codec.h" #include "hda_local.h" +/* define below to restrict the supported rates and formats */ +/* #define LIMITED_RATE_FMT_SUPPORT */ + struct nvhdmi_spec { struct hda_multi_out multiout; @@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = { {} /* terminator */ }; +#ifdef LIMITED_RATE_FMT_SUPPORT +/* support only the safe format and rate */ +#define SUPPORTED_RATES SNDRV_PCM_RATE_48000 +#define SUPPORTED_MAXBPS 16 +#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#else +/* support all rates and formats */ +#define SUPPORTED_RATES \ + (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define SUPPORTED_MAXBPS 24 +#define SUPPORTED_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#endif + /* * Controls */ @@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { .channels_min = 2, .channels_max = 8, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_8ch, @@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .channels_min = 2, .channels_max = 2, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_2ch, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7ed47f6..c08ca66 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -275,7 +275,7 @@ struct alc_spec { struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[5]; /* initialization verbs + const struct hda_verb *init_verbs[10]; /* initialization verbs * don't forget NULL * termination! */ @@ -1332,15 +1332,20 @@ do_sku: * when the external headphone out jack is plugged" */ if (!spec->autocfg.hp_pins[0]) { + hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ if (tmp == 0) - spec->autocfg.hp_pins[0] = porta; + nid = porta; else if (tmp == 1) - spec->autocfg.hp_pins[0] = porte; + nid = porte; else if (tmp == 2) - spec->autocfg.hp_pins[0] = portd; + nid = portd; else return 1; + for (i = 0; i < spec->autocfg.line_outs; i++) + if (spec->autocfg.line_out_pins[i] == nid) + return 1; + spec->autocfg.hp_pins[0] = nid; } alc_init_auto_hp(codec); @@ -1362,7 +1367,7 @@ static void alc_ssid_check(struct hda_codec *codec, } /* - * Fix-up pin default configurations + * Fix-up pin default configurations and add default verbs */ struct alc_pincfg { @@ -1370,9 +1375,14 @@ struct alc_pincfg { u32 val; }; -static void alc_fix_pincfg(struct hda_codec *codec, +struct alc_fixup { + const struct alc_pincfg *pins; + const struct hda_verb *verbs; +}; + +static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_pincfg **pinfix) + const struct alc_fixup *fix) { const struct alc_pincfg *cfg; @@ -1380,9 +1390,14 @@ static void alc_fix_pincfg(struct hda_codec *codec, if (!quirk) return; - cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + fix += quirk->value; + cfg = fix->pins; + if (cfg) { + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + } + if (fix->verbs) + add_verb(codec->spec, fix->verbs); } /* @@ -7927,8 +7942,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { static struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -7947,8 +7963,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = { static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -7960,6 +7977,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9167,7 +9193,8 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc882_targa_automute, }, [ALC883_TARGA_8ch_DIG] = { - .mixers = { alc883_base_mixer, alc883_chmode_mixer }, + .mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer, + alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, alc883_targa_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), @@ -9581,11 +9608,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { { } }; -static const struct alc_pincfg *alc882_pin_fixes[] = { - [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +static const struct alc_fixup alc882_fixups[] = { + [PINFIX_ABIT_AW9D_MAX] = { + .pins = alc882_abit_aw9d_pinfix + }, }; -static struct snd_pci_quirk alc882_pinfix_tbl[] = { +static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), {} }; @@ -9857,7 +9886,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -12648,7 +12677,7 @@ static struct alc_config_preset alc268_presets[] = { .init_hook = alc268_toshiba_automute, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer, + .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, @@ -12830,12 +12859,15 @@ static int patch_alc268(struct hda_codec *codec) unsigned int wcap = get_wcaps(codec, 0x07); int i; + spec->capsrc_nids = alc268_capsrc_nids; /* get type */ wcap = get_wcaps_type(wcap); if (spec->auto_mic || wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); + if (spec->auto_mic) + fixup_automic_adc(codec); if (spec->auto_mic || spec->input_mux->num_items == 1) add_mixer(spec, alc268_capture_nosrc_mixer); else @@ -12845,7 +12877,6 @@ static int patch_alc268(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); add_mixer(spec, alc268_capture_mixer); } - spec->capsrc_nids = alc268_capsrc_nids; /* set default input source */ for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], @@ -13370,7 +13401,8 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", [ALC269_ASUS_EEEPC_P901] = "eeepc-p901", [ALC269_FUJITSU] = "fujitsu", - [ALC269_LIFEBOOK] = "lifebook" + [ALC269_LIFEBOOK] = "lifebook", + [ALC269_AUTO] = "auto", }; static struct snd_pci_quirk alc269_cfg_tbl[] = { @@ -14344,15 +14376,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec) static void alc861_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; - pin = spec->autocfg.hp_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, + if (spec->autocfg.hp_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.hp_pins[0], + PIN_HP, spec->multiout.hp_nid); - pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, + if (spec->autocfg.speaker_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.speaker_pins[0], + PIN_OUT, spec->multiout.dac_nids[0]); } @@ -15145,7 +15178,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), @@ -15538,6 +15571,29 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC660VD_FIX_ASUS_GPIO1 +}; + +/* reset GPIO1 */ +static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = { + {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + { } +}; + +static const struct alc_fixup alc861vd_fixups[] = { + [ALC660VD_FIX_ASUS_GPIO1] = { + .verbs = alc660vd_fix_asus_gpio1_verbs, + }, +}; + +static struct snd_pci_quirk alc861vd_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1), + {} +}; + static int patch_alc861vd(struct hda_codec *codec) { struct alc_spec *spec; @@ -15559,6 +15615,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ err = alc861vd_parse_auto_config(codec); @@ -16839,6 +16897,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), @@ -17132,70 +17191,145 @@ static struct alc_config_preset alc662_presets[] = { * BIOS auto configuration */ +/* convert from MIX nid to DAC */ +static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid) +{ + if (nid == 0x0f) + return 0x02; + else if (nid >= 0x0c && nid <= 0x0e) + return nid - 0x0c + 0x02; + else + return 0; +} + +/* get MIX nid connected to the given pin targeted to DAC */ +static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin, + hda_nid_t dac) +{ + hda_nid_t mix[4]; + int i, num; + + num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix)); + for (i = 0; i < num; i++) { + if (alc662_mix_to_dac(mix[i]) == dac) + return mix[i]; + } + return 0; +} + +/* look for an empty DAC slot */ +static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t srcs[5]; + int i, j, num; + + num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs)); + if (num < 0) + return 0; + for (i = 0; i < num; i++) { + hda_nid_t nid = alc662_mix_to_dac(srcs[i]); + if (!nid) + continue; + for (j = 0; j < spec->multiout.num_dacs; j++) + if (spec->multiout.dac_nids[j] == nid) + break; + if (j >= spec->multiout.num_dacs) + return nid; + } + return 0; +} + +/* fill in the dac_nids table from the parsed pin configuration */ +static int alc662_auto_fill_dac_nids(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + struct alc_spec *spec = codec->spec; + int i; + hda_nid_t dac; + + spec->multiout.dac_nids = spec->private_dac_nids; + for (i = 0; i < cfg->line_outs; i++) { + dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]); + if (!dac) + continue; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + } + return 0; +} + +static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, + hda_nid_t nid, unsigned int chs) +{ + char name[32]; + sprintf(name, "%s Playback Volume", pfx); + return add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); +} + +static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, + hda_nid_t nid, unsigned int chs) +{ + char name[32]; + sprintf(name, "%s Playback Switch", pfx); + return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT)); +} + +#define alc662_add_stereo_vol(spec, pfx, nid) \ + alc662_add_vol_ctl(spec, pfx, nid, 3) +#define alc662_add_stereo_sw(spec, pfx, nid) \ + alc662_add_sw_ctl(spec, pfx, nid, 3) + /* add playback controls from the parsed DAC table */ -static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, +static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - char name[32]; + struct alc_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid; + hda_nid_t nid, mix; int i, err; for (i = 0; i < cfg->line_outs; i++) { - if (!spec->multiout.dac_nids[i]) + nid = spec->multiout.dac_nids[i]; + if (!nid) + continue; + mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid); + if (!mix) continue; - nid = alc880_idx_to_dac(i); if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, - HDA_OUTPUT)); + err = alc662_add_vol_ctl(spec, "Center", nid, 1); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); + err = alc662_add_vol_ctl(spec, "LFE", nid, 2); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x0e, 1, 0, - HDA_INPUT)); + err = alc662_add_sw_ctl(spec, "Center", mix, 1); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, - HDA_INPUT)); + err = alc662_add_sw_ctl(spec, "LFE", mix, 2); if (err < 0) return err; } else { const char *pfx; if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { - if (!cfg->hp_pins) + if (cfg->hp_outs) pfx = "Speaker"; else pfx = "PCM"; } else pfx = chname[i]; - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); + err = alc662_add_vol_ctl(spec, pfx, nid, 3); if (err < 0) return err; if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) pfx = "Speaker"; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i), - 3, 0, HDA_INPUT)); + err = alc662_add_sw_ctl(spec, pfx, mix, 3); if (err < 0) return err; } @@ -17204,54 +17338,38 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, } /* add playback controls for speaker and HP outputs */ -static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, +/* return DAC nid if any new DAC is assigned */ +static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, const char *pfx) { - hda_nid_t nid; + struct alc_spec *spec = codec->spec; + hda_nid_t nid, mix; int err; - char name[32]; if (!pin) return 0; - - if (pin == 0x17) { - /* ALC663 has a mono output pin on 0x17 */ + nid = alc662_look_for_dac(codec, pin); + if (!nid) { + char name[32]; + /* the corresponding DAC is already occupied */ + if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)) + return 0; /* no way */ + /* create a switch only */ sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT)); - return err; + return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); } - if (alc880_is_fixed_pin(pin)) { - nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */ - /* specify the DAC as the extra output */ - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = nid; - else - spec->multiout.extra_out_nid[0] = nid; - /* control HP volume/switch on the output mixer amp */ - nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); - if (err < 0) - return err; - } else if (alc880_is_multi_pin(pin)) { - /* set manual connection */ - /* we have only a switch on HP-out PIN */ - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - return 0; + mix = alc662_dac_to_mix(codec, pin, nid); + if (!mix) + return 0; + err = alc662_add_vol_ctl(spec, pfx, nid, 3); + if (err < 0) + return err; + err = alc662_add_sw_ctl(spec, pfx, mix, 3); + if (err < 0) + return err; + return nid; } /* create playback/capture controls for input pins */ @@ -17260,30 +17378,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, - int dac_idx) + hda_nid_t dac) { + int i, num; + hda_nid_t srcs[4]; + alc_set_pin_output(codec, nid, pin_type); /* need the manual connection? */ - if (alc880_is_multi_pin(nid)) { - struct alc_spec *spec = codec->spec; - int idx = alc880_multi_pin_idx(nid); - snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0, - AC_VERB_SET_CONNECT_SEL, - alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx])); + num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs)); + if (num <= 1) + return; + for (i = 0; i < num; i++) { + if (alc662_mix_to_dac(srcs[i]) != dac) + continue; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i); + return; } } static void alc662_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int pin_type = get_pin_type(spec->autocfg.line_out_type); int i; for (i = 0; i <= HDA_SIDE; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; - int pin_type = get_pin_type(spec->autocfg.line_out_type); if (nid) alc662_auto_set_output_and_unmute(codec, nid, pin_type, - i); + spec->multiout.dac_nids[i]); } } @@ -17293,12 +17416,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin; pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - /* use dac 0 */ - alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + if (pin) + alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, + spec->multiout.hp_nid); pin = spec->autocfg.speaker_pins[0]; if (pin) - alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); + alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, + spec->multiout.extra_out_nid[0]); } #define ALC662_PIN_CD_NID ALC880_PIN_CD_NID @@ -17336,21 +17460,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (!spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ - err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); + err = alc662_auto_fill_dac_nids(codec, &spec->autocfg); if (err < 0) return err; - err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg); + err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg); if (err < 0) return err; - err = alc662_auto_create_extra_out(spec, + err = alc662_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], "Speaker"); if (err < 0) return err; - err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], + if (err) + spec->multiout.extra_out_nid[0] = err; + err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone"); if (err < 0) return err; + if (err) + spec->multiout.hp_nid = err; err = alc662_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e31e53d..66c0876 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_5ST_NO_FP, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_927X_VOLKNOB, STAC_927X_MODELS }; @@ -182,8 +183,8 @@ struct sigmatel_jack { struct sigmatel_mic_route { hda_nid_t pin; - unsigned char mux_idx; - unsigned char dmux_idx; + signed char mux_idx; + signed char dmux_idx; }; struct sigmatel_spec { @@ -864,10 +865,6 @@ static struct hda_verb stac92hd73xx_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, {} @@ -911,6 +908,16 @@ static struct hda_verb d965_core_init[] = { {} }; +static struct hda_verb dell_3st_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* unmute node 0x1b */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -919,6 +926,14 @@ static struct hda_verb stac927x_core_init[] = { {} }; +static struct hda_verb stac927x_volknob_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* enable analog pc beep path */ + {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -1590,8 +1605,8 @@ static unsigned int ref92hd83xxx_pin_configs[10] = { }; static unsigned int dell_s14_pin_configs[10] = { - 0x02214030, 0x02211010, 0x02a19020, 0x01014050, - 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x0221403f, 0x0221101f, 0x02a19020, 0x90170110, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160, 0x40f000f0, 0x40f000f0, }; @@ -1690,6 +1705,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, + "HP dv6", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, @@ -2001,6 +2018,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs, [STAC_DELL_3ST] = dell_3st_pin_configs, [STAC_DELL_BIOS] = NULL, + [STAC_927X_VOLKNOB] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { @@ -2012,6 +2030,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = "5stack-no-fp", [STAC_DELL_3ST] = "dell-3stack", [STAC_DELL_BIOS] = "dell-bios", + [STAC_927X_VOLKNOB] = "volknob", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -2047,6 +2066,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, "Intel D965", STAC_D965_5ST), + /* volume-knob fixes */ + SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB), {} /* terminator */ }; @@ -3471,18 +3492,26 @@ static int set_mic_route(struct hda_codec *codec, break; if (i <= AUTO_PIN_FRONT_MIC) { /* analog pin */ - mic->dmux_idx = 0; i = get_connection_index(codec, spec->mux_nids[0], pin); if (i < 0) return -1; mic->mux_idx = i; + mic->dmux_idx = -1; + if (spec->dmux_nids) + mic->dmux_idx = get_connection_index(codec, + spec->dmux_nids[0], + spec->mux_nids[0]); } else if (spec->dmux_nids) { /* digital pin */ - mic->mux_idx = 0; i = get_connection_index(codec, spec->dmux_nids[0], pin); if (i < 0) return -1; mic->dmux_idx = i; + mic->mux_idx = -1; + if (spec->mux_nids) + mic->mux_idx = get_connection_index(codec, + spec->mux_nids[0], + spec->dmux_nids[0]); } return 0; } @@ -4166,7 +4195,10 @@ static int stac92xx_init(struct hda_codec *codec) stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], AC_PINCTL_OUT_EN); /* fake event to set up pins */ - stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); + if (cfg->hp_pins[0]) + stac_issue_unsol_event(codec, cfg->hp_pins[0]); + else if (cfg->line_out_pins[0]) + stac_issue_unsol_event(codec, cfg->line_out_pins[0]); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); @@ -4556,11 +4588,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec) mic = &spec->ext_mic; else mic = &spec->int_mic; - if (mic->dmux_idx) + if (mic->dmux_idx >= 0) snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, AC_VERB_SET_CONNECT_SEL, mic->dmux_idx); - else + if (mic->mux_idx >= 0) snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0, AC_VERB_SET_CONNECT_SEL, mic->mux_idx); @@ -4688,8 +4720,13 @@ static int stac92xx_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); /* fake event to set up pins again to override cached values */ - if (spec->hp_detect) - stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); + if (spec->hp_detect) { + if (spec->autocfg.hp_pins[0]) + stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); + else if (spec->autocfg.line_out_pins[0]) + stac_issue_unsol_event(codec, + spec->autocfg.line_out_pins[0]); + } return 0; } @@ -5016,7 +5053,7 @@ again: spec->eapd_switch = 1; break; } - if (spec->board_config > STAC_92HD73XX_REF) { + if (spec->board_config != STAC_92HD73XX_REF) { /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; @@ -5066,7 +5103,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->spec = spec; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; - spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; spec->mux_nids = stac92hd83xxx_mux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids); @@ -5242,7 +5278,7 @@ again: stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); - if (spec->board_config > STAC_92HD71BXX_REF) { + if (spec->board_config != STAC_92HD71BXX_REF) { /* GPIO0 = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; @@ -5375,6 +5411,11 @@ again: case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + /* HP dv6 gives the headphone pin as a line-out. Thus we + * need to set hp_detect flag here to force to enable HP + * detection. + */ + spec->hp_detect = 1; break; case STAC_HP_HDX: spec->num_dmics = 1; @@ -5557,14 +5598,17 @@ static int patch_stac927x(struct hda_codec *codec) spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids; + if (spec->board_config != STAC_D965_REF) { + /* GPIO0 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = 0x01; + spec->gpio_dir = spec->gpio_data = 0x01; + } + switch (spec->board_config) { case STAC_D965_3ST: case STAC_D965_5ST: /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01; - spec->gpio_data = 0x01; spec->num_dmics = 0; - spec->init = d965_core_init; break; case STAC_DELL_BIOS: @@ -5583,32 +5627,26 @@ static int patch_stac927x(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: - /* GPIO2 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04; - spec->gpio_data = 0x04; - switch (codec->subsystem_id) { - case 0x1028022f: - /* correct EAPD to be GPIO0 */ - spec->eapd_mask = spec->gpio_mask = 0x01; - spec->gpio_dir = spec->gpio_data = 0x01; - break; - }; + if (codec->subsystem_id != 0x1028022f) { + /* GPIO2 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = 0x04; + spec->gpio_dir = spec->gpio_data = 0x04; + } spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->init = d965_core_init; + spec->init = dell_3st_core_init; spec->dmux_nids = stac927x_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; + case STAC_927X_VOLKNOB: + spec->num_dmics = 0; + spec->init = stac927x_volknob_core_init; + break; default: - if (spec->board_config > STAC_D965_REF) { - /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = 0x01; - spec->gpio_dir = spec->gpio_data = 0x01; - } spec->num_dmics = 0; - spec->init = stac927x_core_init; + break; } spec->num_caps = STAC927X_NUM_CAPS; diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 3756430..6da21a2 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) /* only use basic functionality for now */ - ice->num_total_dacs = 2; /* only PSDOUT0 is connected */ + /* VT1616 6ch codec connected to PSDOUT0 using packed mode */ + ice->num_total_dacs = 6; ice->num_total_adcs = 2; - /* Chaintech AV-710 has another codecs, which need initialization */ - /* initialize WM8728 codec */ + /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4 + (shared with the SPDIF output). Mixer control for this codec + is not yet supported. */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) { for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) wm_put(ice, wm_inits[i], wm_inits[i+1]); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index cecf1ff..d74033a 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index af6e001..10fc92c 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); - return -EBUSY; + return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY; } if (!force && is_pro_rate_locked(ice)) { spin_unlock_irqrestore(&ice->reg_lock, flags); @@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm = pcm; @@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm_ds = pcm; @@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 171ada5..754867e 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "Sony S1XP", .type = AC97_TUNE_INV_EAPD }, + { + .subvendor = 0x104d, + .subdevice = 0x81c0, + .name = "Sony VAIO VGN-T350P", /*AD1981B*/ + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x104d, + .subdevice = 0x81c5, + .name = "Sony VAIO VGN-B1VP", /*AD1981B*/ + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x1043, .subdevice = 0x80f3, diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index 012c010..51afc04 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -86,7 +86,6 @@ struct lx6464es { /* messaging */ spinlock_t msg_lock; /* message spinlock */ - atomic_t send_message_locked; struct lx_rmh rmh; /* configuration */ @@ -95,7 +94,6 @@ struct lx6464es { uint hardware_running[2]; u32 board_sample_rate; /* sample rate read from * board */ - u32 sample_rate; /* our sample rate */ u16 pcm_granularity; /* board blocksize */ /* dma */ diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5812780..3086b75 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -314,98 +314,6 @@ static inline void lx_message_dump(struct lx_rmh *rmh) #define XILINX_POLL_NO_SLEEP 100 #define XILINX_POLL_ITERATIONS 150 -#if 0 /* not used now */ -static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh) -{ - u32 reg = ED_DSP_TIMED_OUT; - int dwloop; - int answer_received; - - if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) { - snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg); - return -EBUSY; - } - - /* write command */ - lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len); - - snd_BUG_ON(atomic_read(&chip->send_message_locked) != 0); - atomic_set(&chip->send_message_locked, 1); - - /* MicoBlaze gogogo */ - lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); - - /* wait for interrupt to answer */ - for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS; ++dwloop) { - answer_received = atomic_read(&chip->send_message_locked); - if (answer_received == 0) - break; - msleep(1); - } - - if (answer_received == 0) { - /* in Debug mode verify Reg_CSM_MR */ - snd_BUG_ON(!(lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)); - - /* command finished, read status */ - if (rmh->dsp_stat == 0) - reg = lx_dsp_reg_read(chip, eReg_CRM1); - else - reg = 0; - } else { - int i; - snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " - "Interrupts disabled?\n"); - - /* attente bit Reg_CSM_MR */ - for (i = 0; i != XILINX_POLL_ITERATIONS; i++) { - if ((lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)) { - if (rmh->dsp_stat == 0) - reg = lx_dsp_reg_read(chip, eReg_CRM1); - else - reg = 0; - goto polling_successful; - } - - if (i > XILINX_POLL_NO_SLEEP) - msleep(1); - } - snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! " - "polling failed\n"); - -polling_successful: - atomic_set(&chip->send_message_locked, 0); - } - - if ((reg & ERROR_VALUE) == 0) { - /* read response */ - if (rmh->stat_len) { - snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1)); - - lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, - rmh->stat_len); - } - } else - snd_printk(KERN_WARNING LXP "lx_message_send: error_value %x\n", - reg); - - /* clear Reg_CSM_MR */ - lx_dsp_reg_write(chip, eReg_CSM, 0); - - switch (reg) { - case ED_DSP_TIMED_OUT: - snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); - return -ETIMEDOUT; - - case ED_DSP_CRASHED: - snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); - return -EAGAIN; - } - - lx_message_dump(rmh); - return 0; -} -#endif /* not used now */ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) { @@ -423,7 +331,7 @@ static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh) /* MicoBlaze gogogo */ lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC); - /* wait for interrupt to answer */ + /* wait for device to answer */ for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS * 1000; ++dwloop) { if (lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR) { if (rmh->dsp_stat == 0) @@ -1175,10 +1083,6 @@ static int lx_interrupt_ack(struct lx6464es *chip, u32 *r_irqsrc, *r_async_escmd = 1; } - if (irqsrc & MASK_SYS_STATUS_CMD_DONE) - /* xilinx command notification */ - atomic_set(&chip->send_message_locked, 0); - if (irq_async) { /* snd_printd("interrupt: async event pending\n"); */ *r_async_pending = 1; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index acfa476..91683a3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; @@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; unsigned long port = chip->port + 0x10 * idx; unsigned char val; int i, change = 0; @@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = }; static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { - .name = "VIA DXS Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 0, + /* .subdevice set later */ + .name = "PCM Playback Volume", .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .count = 4, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, @@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) } else /* Using DXS when PCM emulation is enabled is really weird */ { - /* Standalone DXS controls */ - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); - if (err < 0) - return err; + for (i = 0; i < 4; ++i) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1( + &snd_via8233_dxs_volume_control, chip); + if (!kctl) + return -ENOMEM; + kctl->id.subdevice = i; + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } } } /* select spdif data slot 10/11 */ diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 835fa19..d06f780 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) strlcpy(info.type, "keywest", I2C_NAME_SIZE); info.addr = keywest_ctx->addr; keywest_ctx->client = i2c_new_device(adapter, &info); + if (!keywest_ctx->client) + return -ENODEV; + /* + * We know the driver is already loaded, so the device should be + * already bound. If not it means binding failed, and then there + * is no point in keeping the device instantiated. + */ + if (!keywest_ctx->client->driver) { + i2c_unregister_device(keywest_ctx->client); + keywest_ctx->client = NULL; + return -ENODEV; + } /* * Let i2c-core delete that device on driver removal. @@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = { { } }; -struct i2c_driver keywest_driver = { +static struct i2c_driver keywest_driver = { .driver = { .name = "PMac Keywest Audio", }, diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index ac927ff..97f1a25 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,15 +7,6 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. -config SND_BF5XX_TDM - tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" - depends on (BLACKFIN && SND_SOC) - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in TDM - mode. - You will also need to select the audio interfaces to support below. - config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE Enter the GPIO used to control AD73311's SE pin. Acceptable values are 0 to 7 +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + +config SND_BF5XX_SOC_AD1938 + tristate "SoC AD1938 Audio support for Blackfin" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1938 + help + Say Y if you want to add support for AD1938 codec on Blackfin. + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN @@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT Say y if you want AC97 driver to support up to 5.1 channel audio. this mode will consume much more memory for DMA. +config SND_BF5XX_HAVE_COLD_RESET + bool "BOARD has COLD Reset GPIO" + depends on SND_BF5XX_AC97 + default y if BFIN548_EZKIT + default n if !BFIN548_EZKIT + +config SND_BF5XX_RESET_GPIO_NUM + int "Set a GPIO for cold reset" + depends on SND_BF5XX_HAVE_COLD_RESET + range 0 159 + default 19 if BFIN548_EZKIT + default 5 if BFIN537_STAMP + default 0 + help + Set the correct GPIO for RESET the sound chip. + +config SND_BF5XX_SOC_AD1980 + tristate "SoC AD1980/1 Audio support for BF5xx" + depends on SND_BF5XX_AC97 + select SND_BF5XX_SOC_AC97 + select SND_SOC_AD1980 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_SPORT tristate @@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx" - depends on SND_BF5XX_AC97 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" - depends on SND_BF5XX_TDM - select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 - help - Say Y if you want to add support for AD1938 codec on Blackfin. - config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) @@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM default 0 help Set the correct SPORT for sound chip. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 2758b90..e693229 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); +#else ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); +#endif if (ret) { pr_err("SPORT is busy!\n"); ret = -EBUSY; diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f2a911..a1f97dd 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-ac97.h + * sound/soc/blackfin/bf5xx-ac97.h * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 876abad..084b688 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = { * TFS. When Port G is selected and EMAC then there is a conflict between * the PHY interrupt line and TFS. Current settings prevent the conflict * by ignoring the TFS pin when Port G is selected. This allows both - * ssm2602 using Port G and EMAC concurrently. + * codecs and EMAC using Port G concurrently. */ -#ifdef CONFIG_BF527_SPORT0_PORTF -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#else +#ifdef CONFIG_BF527_SPORT0_PORTG #define LOCAL_SPORT0_TFS (0) +#else +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) #endif static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, @@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } -static void bf5xx_i2s_remove(struct snd_soc_dai *dai) +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); peripheral_free_list(&sport_req[sport_num][0]); @@ -236,36 +237,31 @@ static void bf5xx_i2s_remove(struct snd_soc_dai *dai) #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; + if (dai->capture.active) - sport_rx_stop(sport); + sport_rx_stop(sport_handle); if (dai->playback.active) - sport_tx_stop(sport); + sport_tx_stop(sport_handle); return 0; } static int bf5xx_i2s_resume(struct snd_soc_dai *dai) { int ret; - struct sport_device *sport = - (struct sport_device *)dai->private_data; pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h index 7107d1a..264ecdc 100644 --- a/sound/soc/blackfin/bf5xx-i2s.h +++ b/sound/soc/blackfin/bf5xx-i2s.h @@ -1,5 +1,5 @@ /* - * linux/sound/arm/bf5xx-i2s.h + * sound/soc/blackfin/bf5xx-i2s.h * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 469ce7f..99051ff 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) int sport_tx_start(struct sport_device *sport) { - unsigned flags; + unsigned long flags; pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, sport->tx_run, sport->rx_run); if (sport->tx_run) diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..ff546e9 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = { * TFS. When Port G is selected and EMAC then there is a conflict between * the PHY interrupt line and TFS. Current settings prevent the conflict * by ignoring the TFS pin when Port G is selected. This allows both - * ssm2602 using Port G and EMAC concurrently. + * codecs and EMAC using Port G concurrently. */ -#ifdef CONFIG_BF527_SPORT0_PORTF -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#else +#ifdef CONFIG_BF527_SPORT0_PORTG #define LOCAL_SPORT0_TFS (0) +#else +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) #endif static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3612bb9..c48485f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -18,7 +18,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> @@ -252,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index e62b277..34b30ef 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -28,7 +28,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> @@ -457,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -516,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3ff0373..593d5b9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_LEFT_INPUT_VOLUME, 14, 1, 0), + WM8350_LEFT_INPUT_VOLUME, 14, 1, 1), }; /* Right Input Mixer */ @@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0), + WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1), }; /* Left Mic Mixer */ diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414..5ad677c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"}, diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae..1ef2454 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index d8a013a..98d663a 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c64e55a..686e5aa 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1027,7 +1027,7 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, - wm9081->fs); for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { cur_val = abs((wm9081->sysclk_rate / - clk_sys_rates[i].ratio) - wm9081->fs);; + clk_sys_rates[i].ratio) - wm9081->fs); if (cur_val < best_val) { best = i; best_val = cur_val; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 12a6c54..4ae7070 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,22 +97,19 @@ enum { DAVINCI_MCBSP_WORD_32, }; -static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { - .name = "I2S PCM Stereo out", -}; - -static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { - .name = "I2S PCM Stereo in", -}; - struct davinci_mcbsp_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to a pointer of an array of struct davinci_pcm_dma_params in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 int mode; u32 pcr; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct davinci_pcm_dma_params *dma_params = dai->dma_data; struct davinci_mcbsp_dev *dev = dai->private_data; + struct davinci_pcm_dma_params *dma_params = + &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; @@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->base = (void __iomem *)IO_ADDRESS(mem->start); - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_free_mem; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_free_mem; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; ret = snd_soc_register_dai(&davinci_i2s_dai); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index eca22d7..5d1f98a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) printk(KERN_ERR "GBLCTL write error\n"); } -static int davinci_mcasp_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct davinci_audio_dev *dev = cpu_dai->private_data; - cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; -} - static void mcasp_start_rx(struct davinci_audio_dev *dev) { mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); @@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_start_tx(dev); - else + } else { + if (dev->rxnumevt) /* enable FIFO */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_start_rx(dev); - - /* enable FIFO */ - if (dev->txnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev) static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + FIFO_ENABLE); mcasp_stop_tx(dev); - else + } else { + if (dev->rxnumevt) /* disable FIFO */ + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + FIFO_ENABLE); mcasp_stop_rx(dev); - - /* disable FIFO */ - if (dev->txnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); - - if (dev->rxnumevt) - mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -512,34 +504,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int channel_size) { u32 fmt = 0; + u32 mask, rotate; switch (channel_size) { case DAVINCI_AUDIO_WORD_8: fmt = 0x03; + rotate = 6; + mask = 0x000000ff; break; case DAVINCI_AUDIO_WORD_12: fmt = 0x05; + rotate = 5; + mask = 0x00000fff; break; case DAVINCI_AUDIO_WORD_16: fmt = 0x07; + rotate = 4; + mask = 0x0000ffff; break; case DAVINCI_AUDIO_WORD_20: fmt = 0x09; + rotate = 3; + mask = 0x000fffff; break; case DAVINCI_AUDIO_WORD_24: fmt = 0x0B; + rotate = 2; + mask = 0x00ffffff; break; case DAVINCI_AUDIO_WORD_28: fmt = 0x0D; + rotate = 1; + mask = 0x0fffffff; break; case DAVINCI_AUDIO_WORD_32: fmt = 0x0F; + rotate = 0; + mask = 0xffffffff; break; default: @@ -550,6 +557,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, RXSSZ(fmt), RXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate), + TXROT(7)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate), + RXROT(7)); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); + return 0; } @@ -638,7 +652,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "playback tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); } else { /* bit stream is MSB first with no delay */ @@ -655,7 +668,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) printk(KERN_ERR "capture tdm slot %d not supported\n", dev->tdm_slots); - mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); } } @@ -700,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, { struct davinci_audio_dev *dev = cpu_dai->private_data; struct davinci_pcm_dma_params *dma_params = - dev->dma_params[substream->stream]; + &dev->dma_params[substream->stream]; int word_length; u8 numevt; @@ -778,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { - .startup = davinci_mcasp_startup, .trigger = davinci_mcasp_trigger, .hw_params = davinci_mcasp_hw_params, .set_fmt = davinci_mcasp_set_dai_fmt, @@ -829,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res; struct snd_platform_data *pdata; struct davinci_audio_dev *dev; - int count = 0; int ret = 0; dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); if (!dev) return -ENOMEM; - dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2, - GFP_KERNEL); - if (!dma_data) { - ret = -ENOMEM; - goto err_release_dev; - } - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "no mem resource?\n"); @@ -877,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->txnumevt = pdata->txnumevt; dev->rxnumevt = pdata->rxnumevt; - dma_data[count].name = "I2S PCM Stereo out"; - dma_data[count].eventq_no = pdata->eventq_no; - dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->eventq_no = pdata->eventq_no; + dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + io_v2p(dev->base)); - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count]; /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -890,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err_release_region; } - dma_data[count].channel = res->start; - count++; - dma_data[count].name = "I2S PCM Stereo in"; - dma_data[count].eventq_no = pdata->eventq_no; - dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + dma_data->channel = res->start; + + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->eventq_no = pdata->eventq_no; + dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + io_v2p(dev->base)); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -904,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err_release_region; } - dma_data[count].channel = res->start; + dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); @@ -916,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev) err_release_region: release_mem_region(mem->start, (mem->end - mem->start) + 1); err_release_data: - kfree(dma_data); -err_release_dev: kfree(dev); return ret; @@ -926,7 +925,6 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { struct snd_platform_data *pdata = pdev->dev.platform_data; - struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; @@ -939,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev) mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); - dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - kfree(dma_data); kfree(dev); return 0; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 554354c..9d179cc 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,10 +39,15 @@ enum { }; struct davinci_audio_dev { + /* + * dma_params must be first because rtd->dai->cpu_dai->private_data + * is cast to a pointer of an array of struct davinci_pcm_dma_params in + * davinci_pcm_open. + */ + struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; struct clk *clk; - struct davinci_pcm_dma_params *dma_params[2]; unsigned int codec_fmt; /* McASP specific data */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 091dacb..c73a915 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; struct edmacc_param p_ram; int ret; - if (!dma_data) - return -ENODEV; - - prtd->params = dma_data; - /* Request master DMA channel */ ret = edma_alloc_channel(prtd->params->channel, davinci_pcm_dma_irq, substream, @@ -145,7 +138,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) prtd->master_lch = ret; /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_SLOT_ANY); + ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); if (ret < 0) { edma_free_channel(prtd->master_lch); return ret; @@ -162,8 +155,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch); - p_ram.link_bcntrld = prtd->slave_lch << 5; + p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); + p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; edma_write_slot(prtd->slave_lch, &p_ram); return 0; @@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *params = &pa[substream->stream]; + if (!params) + return -ENODEV; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); /* ensure that buffer size is a multiple of period size */ @@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; spin_lock_init(&prtd->lock); + prtd->params = params; runtime->private_data = prtd; diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 63d9625..8746606 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -17,7 +17,6 @@ struct davinci_pcm_dma_params { - char *name; /* stream identifier */ int channel; /* sync dma channel ID */ unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..6096d22 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op) int size, irq, rc; const __be32 *prop; void __iomem *regs; + int ret; /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); @@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { - iounmap(regs); - return -ENOMEM; + ret = -ENOMEM; + goto out_unmap; } /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; + if (!prop || size < sizeof *prop) { + ret = -ENODEV; + goto out_free; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex); @@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op) if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + ret = -ENODEV; + goto out_free; } /* Disable all interrupts and reset the PSC */ @@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op) &psc_dma_bcom_irq_tx, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; + ret = -ENODEV; + goto out_irq; } /* Save what we've done so it can be found again later */ @@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Tell the ASoC OF helpers about it */ return snd_soc_register_platform(&mpc5200_audio_dma_platform); +out_irq: + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); +out_free: + kfree(psc_dma); +out_unmap: + iounmap(regs); + return ret; } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create); diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c index 3806ff2..ccdefe6 100644 --- a/sound/soc/imx/mxc-ssi.c +++ b/sound/soc/imx/mxc-ssi.c @@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; } - /* sync */ - if (!(fmt & SND_SOC_DAIFMT_ASYNC)) - scr |= SSI_SCR_SYN; - - /* tdm - only for stereo atm */ - if (fmt & SND_SOC_DAIFMT_TDM) - scr |= SSI_SCR_NET; - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { SSI1_STCR = stcr; SSI1_SRCR = srcr; diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 6375b4e..dcb3181 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701 config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5b9ed64..d11a6d7 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -351,7 +351,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, do_div(tmp, freq_out); val = tmp; - val = (val << 16) | 64;; + val = (val << 16) | 64; ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index aa7af0b..9bc4aa3 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + /* * Wait for the LR signal to allow synchronisation to the L/R clock * from the codec. May only be needed for slave mode. @@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + unsigned long loops = msecs_to_loops(5); pr_debug("Entered %s\n", __func__); - while (1) { + while (--loops) { iiscon = readl(i2s->regs + S3C2412_IISCON); if (iiscon & S3C2412_IISCON_LRINDEX) break; - if (timeout < jiffies) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; } return 0; diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index 8e79a41..c215d32 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -67,7 +67,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { int ret = 0; #ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime;; + struct snd_pcm_runtime *runtime = substream->runtime; #endif mutex_lock(&clk_lock); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d8b08e..8de6f9d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) /* connected jack or spk ? */ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - widget->id == snd_soc_dapm_line) + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) return 1; } @@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 1; /* connected jack ? */ - if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line) + if (widget->id == snd_soc_dapm_mic || + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) return 1; } @@ -1131,9 +1132,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file, ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n", w->name, w->power ? "On" : "Off", in, out); - if (w->active && w->sname) - ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n", - w->sname); + if (w->sname) + ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + w->sname, + w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { if (p->connect) diff --git a/sound/sound_core.c b/sound/sound_core.c index bb4b88e..49c9981 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -29,7 +29,7 @@ MODULE_DESCRIPTION("Core sound module"); MODULE_AUTHOR("Alan Cox"); MODULE_LICENSE("GPL"); -static char *sound_nodename(struct device *dev) +static char *sound_devnode(struct device *dev, mode_t *mode) { if (MAJOR(dev->devt) == SOUND_MAJOR) return NULL; @@ -50,7 +50,7 @@ static int __init init_soundcore(void) return PTR_ERR(sound_class); } - sound_class->nodename = sound_nodename; + sound_class->devnode = sound_devnode; return 0; } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ab5a3ac..9efcfd0 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = { * build a feature control */ +static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) +{ + return strlcat(kctl->id.name, str, sizeof(kctl->id.name)); +} + static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, unsigned int ctl_mask, int control, struct usb_audio_term *iterm, int unitid) @@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, */ if (! mapped_name && ! (state->oterm.type >> 16)) { if ((state->oterm.type & 0xff00) == 0x0100) { - len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name)); + len = append_ctl_name(kctl, " Capture"); } else { - len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name)); + len = append_ctl_name(kctl, " Playback"); } } - strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume", - sizeof(kctl->id.name)); + append_ctl_name(kctl, control == USB_FEATURE_MUTE ? + " Switch" : " Volume"); if (control == USB_FEATURE_VOLUME) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= @@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); if (! len) len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1); - strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name)); + append_ctl_name(kctl, " Volume"); snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n", cval->id, kctl->id.name, cval->channels, cval->min, cval->max); @@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned if (! len) strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); } - strlcat(kctl->id.name, " ", sizeof(kctl->id.name)); - strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name)); + append_ctl_name(kctl, " "); + append_ctl_name(kctl, valinfo->suffix); snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n", cval->id, kctl->id.name, cval->channels, cval->min, cval->max); @@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); if ((state->oterm.type & 0xff00) == 0x0100) - strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name)); + append_ctl_name(kctl, " Capture Source"); else - strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name)); + append_ctl_name(kctl, " Playback Source"); } snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n", diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index fd44946..99f3376 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -154,7 +154,7 @@ static void usb_stream_hwdep_vm_close(struct vm_area_struct *area) snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count)); } -static struct vm_operations_struct usb_stream_hwdep_vm_ops = { +static const struct vm_operations_struct usb_stream_hwdep_vm_ops = { .open = usb_stream_hwdep_vm_open, .fault = usb_stream_hwdep_vm_fault, .close = usb_stream_hwdep_vm_close, diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index f3d8f71..52e04b2 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -53,7 +53,7 @@ static int snd_us428ctls_vm_fault(struct vm_area_struct *area, return 0; } -static struct vm_operations_struct us428ctls_vm_ops = { +static const struct vm_operations_struct us428ctls_vm_ops = { .fault = snd_us428ctls_vm_fault, }; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 117946f..4b2304c 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -697,7 +697,7 @@ static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_area_struct *area, } -static struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = { +static const struct vm_operations_struct snd_usX2Y_hwdep_pcm_vm_ops = { .open = snd_usX2Y_hwdep_pcm_vm_open, .close = snd_usX2Y_hwdep_pcm_vm_close, .fault = snd_usX2Y_hwdep_pcm_vm_fault, |