diff options
Diffstat (limited to 'sound')
100 files changed, 2633 insertions, 2313 deletions
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e49f448ee04f..c2db7e905f7d 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -455,7 +455,6 @@ static int snd_pcm_hw_param_near(struct snd_pcm_substream *pcm, v = snd_pcm_hw_param_last(pcm, params, var, dir); else v = snd_pcm_hw_param_first(pcm, params, var, dir); - snd_BUG_ON(v < 0); return v; } @@ -1335,8 +1334,11 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { tmp = bytes; if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes) @@ -1380,14 +1382,18 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha xfer += tmp; if ((substream->f_flags & O_NONBLOCK) != 0 && tmp != runtime->oss.period_bytes) - break; + tmp = -EAGAIN; } - } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - err: - mutex_unlock(&runtime->oss.params_lock); + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + break; + } + tmp = 0; + } return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } @@ -1435,8 +1441,11 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) return tmp; - mutex_lock(&runtime->oss.params_lock); while (bytes > 0) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + tmp = -ERESTARTSYS; + break; + } if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { if (runtime->oss.buffer_used == 0) { tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1); @@ -1467,12 +1476,16 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use bytes -= tmp; xfer += tmp; } - } - mutex_unlock(&runtime->oss.params_lock); - return xfer; - err: - mutex_unlock(&runtime->oss.params_lock); + mutex_unlock(&runtime->oss.params_lock); + if (tmp < 0) + break; + if (signal_pending(current)) { + tmp = -ERESTARTSYS; + break; + } + tmp = 0; + } return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index cadc93792868..85a56af104bd 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -592,18 +592,26 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st snd_pcm_sframes_t frames = size; plugin = snd_pcm_plug_first(plug); - while (plugin && frames > 0) { + while (plugin) { + if (frames <= 0) + return frames; if ((next = plugin->next) != NULL) { snd_pcm_sframes_t frames1 = frames; - if (plugin->dst_frames) + if (plugin->dst_frames) { frames1 = plugin->dst_frames(plugin, frames); + if (frames1 <= 0) + return frames1; + } if ((err = next->client_channels(next, frames1, &dst_channels)) < 0) { return err; } if (err != frames1) { frames = err; - if (plugin->src_frames) + if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames1); + if (frames <= 0) + return frames; + } } } else dst_channels = NULL; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9070f277f8db..09ee8c6b9f75 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -153,7 +153,9 @@ static int snd_pcm_control_ioctl(struct snd_card *card, err = -ENXIO; goto _error; } + mutex_lock(&pcm->open_mutex); err = snd_pcm_info_user(substream, info); + mutex_unlock(&pcm->open_mutex); _error: mutex_unlock(®ister_mutex); return err; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 10e7ef7a8804..db7894bb028c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1632,7 +1632,7 @@ int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); @@ -1678,7 +1678,7 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, return changed; if (params->rmask) { int err = snd_pcm_hw_refine(pcm, params); - if (snd_BUG_ON(err < 0)) + if (err < 0) return err; } return snd_pcm_hw_param_value(params, var, dir); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a4d92e46c459..f08772568c17 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2580,7 +2580,7 @@ static snd_pcm_sframes_t forward_appl_ptr(struct snd_pcm_substream *substream, return ret < 0 ? ret : frames; } -/* decrease the appl_ptr; returns the processed frames or a negative error */ +/* decrease the appl_ptr; returns the processed frames or zero for error */ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream, snd_pcm_uframes_t frames, snd_pcm_sframes_t avail) @@ -2597,7 +2597,12 @@ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream, if (appl_ptr < 0) appl_ptr += runtime->boundary; ret = pcm_lib_apply_appl_ptr(substream, appl_ptr); - return ret < 0 ? ret : frames; + /* NOTE: we return zero for errors because PulseAudio gets depressed + * upon receiving an error from rewind ioctl and stops processing + * any longer. Returning zero means that no rewind is done, so + * it's not absolutely wrong to answer like that. + */ + return ret < 0 ? 0 : frames; } static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream, diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b3b353d72527..f055ca10bbc1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -579,15 +579,14 @@ static int snd_rawmidi_info_user(struct snd_rawmidi_substream *substream, return 0; } -int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +static int __snd_rawmidi_info_select(struct snd_card *card, + struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; struct snd_rawmidi_str *pstr; struct snd_rawmidi_substream *substream; - mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, info->device); - mutex_unlock(®ister_mutex); if (!rmidi) return -ENXIO; if (info->stream < 0 || info->stream > 1) @@ -603,6 +602,16 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info } return -ENXIO; } + +int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info) +{ + int ret; + + mutex_lock(®ister_mutex); + ret = __snd_rawmidi_info_select(card, info); + mutex_unlock(®ister_mutex); + return ret; +} EXPORT_SYMBOL(snd_rawmidi_info_select); static int snd_rawmidi_info_select_user(struct snd_card *card, diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 37d9cfbc29f9..b80985fbc334 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -355,7 +355,7 @@ static int initialize_timer(struct snd_seq_timer *tmr) unsigned long freq; t = tmr->timeri->timer; - if (snd_BUG_ON(!t)) + if (!t) return -EINVAL; freq = tmr->preferred_resolution; diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index afac886ffa28..0333143a1fa7 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -39,6 +39,7 @@ #include <sound/core.h> #include <sound/control.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/info.h> #include <sound/initval.h> @@ -305,19 +306,6 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static void params_change_substream(struct loopback_pcm *dpcm, - struct snd_pcm_runtime *runtime) -{ - struct snd_pcm_runtime *dst_runtime; - - if (dpcm == NULL || dpcm->substream == NULL) - return; - dst_runtime = dpcm->substream->runtime; - if (dst_runtime == NULL) - return; - dst_runtime->hw = dpcm->cable->hw; -} - static void params_change(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -329,10 +317,6 @@ static void params_change(struct snd_pcm_substream *substream) cable->hw.rate_max = runtime->rate; cable->hw.channels_min = runtime->channels; cable->hw.channels_max = runtime->channels; - params_change_substream(cable->streams[SNDRV_PCM_STREAM_PLAYBACK], - runtime); - params_change_substream(cable->streams[SNDRV_PCM_STREAM_CAPTURE], - runtime); } static int loopback_prepare(struct snd_pcm_substream *substream) @@ -620,26 +604,29 @@ static unsigned int get_cable_index(struct snd_pcm_substream *substream) static int rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; + struct snd_mask m; - struct snd_pcm_hardware *hw = rule->private; - struct snd_mask *maskp = hw_param_mask(params, rule->var); - - maskp->bits[0] &= (u_int32_t)hw->formats; - maskp->bits[1] &= (u_int32_t)(hw->formats >> 32); - memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ - if (! maskp->bits[0] && ! maskp->bits[1]) - return -EINVAL; - return 0; + snd_mask_none(&m); + mutex_lock(&dpcm->loopback->cable_lock); + m.bits[0] = (u_int32_t)cable->hw.formats; + m.bits[1] = (u_int32_t)(cable->hw.formats >> 32); + mutex_unlock(&dpcm->loopback->cable_lock); + return snd_mask_refine(hw_param_mask(params, rule->var), &m); } static int rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->rate_min; - t.max = hw->rate_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.rate_min; + t.max = cable->hw.rate_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); @@ -648,22 +635,44 @@ static int rule_rate(struct snd_pcm_hw_params *params, static int rule_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hardware *hw = rule->private; + struct loopback_pcm *dpcm = rule->private; + struct loopback_cable *cable = dpcm->cable; struct snd_interval t; - t.min = hw->channels_min; - t.max = hw->channels_max; + mutex_lock(&dpcm->loopback->cable_lock); + t.min = cable->hw.channels_min; + t.max = cable->hw.channels_max; + mutex_unlock(&dpcm->loopback->cable_lock); t.openmin = t.openmax = 0; t.integer = 0; return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +static void free_cable(struct snd_pcm_substream *substream) +{ + struct loopback *loopback = substream->private_data; + int dev = get_cable_index(substream); + struct loopback_cable *cable; + + cable = loopback->cables[substream->number][dev]; + if (!cable) + return; + if (cable->streams[!substream->stream]) { + /* other stream is still alive */ + cable->streams[substream->stream] = NULL; + } else { + /* free the cable */ + loopback->cables[substream->number][dev] = NULL; + kfree(cable); + } +} + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm; - struct loopback_cable *cable; + struct loopback_cable *cable = NULL; int err = 0; int dev = get_cable_index(substream); @@ -681,7 +690,6 @@ static int loopback_open(struct snd_pcm_substream *substream) if (!cable) { cable = kzalloc(sizeof(*cable), GFP_KERNEL); if (!cable) { - kfree(dpcm); err = -ENOMEM; goto unlock; } @@ -699,19 +707,19 @@ static int loopback_open(struct snd_pcm_substream *substream) /* are cached -> they do not reflect the actual state */ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - rule_format, &runtime->hw, + rule_format, dpcm, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rule_rate, &runtime->hw, + rule_rate, dpcm, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) goto unlock; err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rule_channels, &runtime->hw, + rule_channels, dpcm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) goto unlock; @@ -723,6 +731,10 @@ static int loopback_open(struct snd_pcm_substream *substream) else runtime->hw = cable->hw; unlock: + if (err < 0) { + free_cable(substream); + kfree(dpcm); + } mutex_unlock(&loopback->cable_lock); return err; } @@ -731,20 +743,10 @@ static int loopback_close(struct snd_pcm_substream *substream) { struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm = substream->runtime->private_data; - struct loopback_cable *cable; - int dev = get_cable_index(substream); loopback_timer_stop(dpcm); mutex_lock(&loopback->cable_lock); - cable = loopback->cables[substream->number][dev]; - if (cable->streams[!substream->stream]) { - /* other stream is still alive */ - cable->streams[substream->stream] = NULL; - } else { - /* free the cable */ - loopback->cables[substream->number][dev] = NULL; - kfree(cable); - } + free_cable(substream); mutex_unlock(&loopback->cable_lock); return 0; } diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 038a180d3f81..cbe818eda336 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -325,7 +325,7 @@ static int hdac_component_master_match(struct device *dev, void *data) */ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops) { - if (WARN_ON(!hdac_acomp)) + if (!hdac_acomp) return -ENODEV; hdac_acomp->audio_ops = aops; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a81aacf684b2..37e1cf8218ff 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -271,6 +271,8 @@ enum { CXT_FIXUP_HP_SPECTRE, CXT_FIXUP_HP_GATE_MIC, CXT_FIXUP_MUTE_LED_GPIO, + CXT_FIXUP_HEADSET_MIC, + CXT_FIXUP_HP_MIC_NO_PRESENCE, }; /* for hda_fixup_thinkpad_acpi() */ @@ -350,6 +352,18 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec, } } +static void cxt_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + break; + } +} + /* OPLC XO 1.5 fixup */ /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) @@ -880,6 +894,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_mute_led_gpio, }, + [CXT_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headset_mic, + }, + [CXT_FIXUP_HP_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02a1113c }, + { } + }, + .chained = true, + .chain_id = CXT_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -934,6 +961,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c19c81d230bd..b4f1b6e88305 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -55,10 +55,11 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_kabylake(codec) ((codec)->core.vendor_id == 0x8086280b) #define is_geminilake(codec) (((codec)->core.vendor_id == 0x8086280d) || \ ((codec)->core.vendor_id == 0x80862800)) +#define is_cannonlake(codec) ((codec)->core.vendor_id == 0x8086280c) #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec) \ || is_skylake(codec) || is_broxton(codec) \ - || is_kabylake(codec)) || is_geminilake(codec) - + || is_kabylake(codec)) || is_geminilake(codec) \ + || is_cannonlake(codec) #define is_valleyview(codec) ((codec)->core.vendor_id == 0x80862882) #define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883) #define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) @@ -3841,6 +3842,7 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862800, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 921a10eff43a..8fd2d9c62c96 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -324,21 +324,24 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0292: alc_update_coef_idx(codec, 0x4, 1<<15, 0); break; - case 0x10ec0215: case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); + /* fallthrough */ + case 0x10ec0215: case 0x10ec0233: case 0x10ec0236: case 0x10ec0255: case 0x10ec0256: + case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: case 0x10ec0285: - case 0x10ec0295: case 0x10ec0298: case 0x10ec0289: - case 0x10ec0299: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; case 0x10ec0275: @@ -2772,6 +2775,7 @@ enum { ALC269_TYPE_ALC298, ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, + ALC269_TYPE_ALC257, ALC269_TYPE_ALC215, ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, @@ -2805,6 +2809,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC298: case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: + case ALC269_TYPE_ALC257: case ALC269_TYPE_ALC215: case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: @@ -5182,6 +5187,22 @@ static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, } } +/* Forcibly assign NID 0x03 to HP/LO while NID 0x02 to SPK for EQ */ +static void alc274_fixup_bind_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t preferred_pairs[] = { + 0x21, 0x03, 0x1b, 0x03, 0x16, 0x02, + 0 + }; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.preferred_dacs = preferred_pairs; +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -5299,6 +5320,8 @@ enum { ALC233_FIXUP_LENOVO_MULTI_CODECS, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC700_FIXUP_INTEL_REFERENCE, + ALC274_FIXUP_DELL_BIND_DACS, + ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, }; static const struct hda_fixup alc269_fixups[] = { @@ -6109,6 +6132,21 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC274_FIXUP_DELL_BIND_DACS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_bind_dacs, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, + [ALC274_FIXUP_DELL_AIO_LINEOUT_VERB] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x0401102f }, + { } + }, + .chained = true, + .chain_id = ALC274_FIXUP_DELL_BIND_DACS + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6292,6 +6330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), @@ -6550,6 +6589,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01011020}, {0x21, 0x02211010}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x1b, 0x01011020}, + {0x21, 0x0221101f}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x21, 0x02211030}), @@ -6575,7 +6619,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x12, 0xb7a60130}, {0x13, 0xb8a61140}, {0x16, 0x90170110}, @@ -6867,6 +6911,10 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; + case 0x10ec0257: + spec->codec_variant = ALC269_TYPE_ALC257; + spec->gen.mixer_nid = 0; + break; case 0x10ec0215: case 0x10ec0285: case 0x10ec0289: @@ -7914,6 +7962,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260), HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262), HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268), diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9f521a55d610..c33a512283a4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -850,6 +850,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -875,6 +878,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1051,6 +1056,11 @@ static int acp_audio_probe(struct platform_device *pdev) struct resource *res; const u32 *pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "Missing platform data\n"); + return -ENODEV; + } + audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); if (audio_drv_data == NULL) @@ -1058,6 +1068,8 @@ static int acp_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audio_drv_data->acp_mmio = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audio_drv_data->acp_mmio)) + return PTR_ERR(audio_drv_data->acp_mmio); /* The following members gets populated in device 'open' * function. Till then interrupts are disabled in 'acp_init' @@ -1084,7 +1096,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed status:%d\n", status); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1101,9 +1117,12 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) + dev_err(&pdev->dev, "ACP Deinit failed status:%d\n", status); snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1113,9 +1132,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed status:%d\n", status); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1147,18 +1171,26 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) + dev_err(dev, "ACP Deinit failed status:%d\n", status); acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed status:%d\n", status); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba51..dcee145dd179 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -64,7 +64,7 @@ config SND_AT91_SOC_SAM9X5_WM8731 config SND_ATMEL_SOC_CLASSD tristate "Atmel ASoC driver for boards using CLASSD" depends on ARCH_AT91 || COMPILE_TEST - select SND_ATMEL_SOC_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM select REGMAP_MMIO help Say Y if you want to add support for Atmel ASoC driver for boards using diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 8445edd06737..ebabed69f0e6 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -308,15 +308,9 @@ static int atmel_classd_codec_resume(struct snd_soc_codec *codec) return regcache_sync(dd->regmap); } -static struct regmap *atmel_classd_codec_get_remap(struct device *dev) -{ - return dev_get_regmap(dev, NULL); -} - static struct snd_soc_codec_driver soc_codec_dev_classd = { .probe = atmel_classd_codec_probe, .resume = atmel_classd_codec_resume, - .get_regmap = atmel_classd_codec_get_remap, .component_driver = { .controls = atmel_classd_snd_controls, .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 29a97d52e8ad..66d6c52e7761 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -91,8 +91,8 @@ static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, do { mutex_lock(&ctx->lock); - tmo = 5; - while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + tmo = 6; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && --tmo) udelay(21); /* wait an ac97 frame time */ if (!tmo) { pr_debug("ac97rd timeout #1\n"); @@ -105,7 +105,7 @@ static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, * poll, Forrest, poll... */ tmo = 0x10000; - while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + while ((RD(ctx, AC97_STATUS) & STAT_CP) && --tmo) asm volatile ("nop"); data = RD(ctx, AC97_CMDRESP); diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 2e449d7173fc..d5f73a8ab893 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -130,6 +130,7 @@ struct bcm2835_i2s_dev { struct regmap *i2s_regmap; struct clk *clk; bool clk_prepared; + int clk_rate; }; static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) @@ -419,10 +420,19 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, } /* Clock should only be set up here if CPU is clock master */ - if (bit_clock_master) { - ret = clk_set_rate(dev->clk, bclk_rate); - if (ret) - return ret; + if (bit_clock_master && + (!dev->clk_prepared || dev->clk_rate != bclk_rate)) { + if (dev->clk_prepared) + bcm2835_i2s_stop_clock(dev); + + if (dev->clk_rate != bclk_rate) { + ret = clk_set_rate(dev->clk, bclk_rate); + if (ret) + return ret; + dev->clk_rate = bclk_rate; + } + + bcm2835_i2s_start_clock(dev); } /* Setup the frame format */ @@ -618,8 +628,6 @@ static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, struct bcm2835_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); uint32_t cs_reg; - bcm2835_i2s_start_clock(dev); - /* * Clear both FIFOs if the one that should be started * is not empty at the moment. This should only happen diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 848c5fe49bc7..be8ea723dff9 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1319,6 +1319,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) int i, ret; pm860x->codec = codec; + snd_soc_codec_init_regmap(codec, pm860x->regmap); for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, @@ -1348,18 +1349,10 @@ static int pm860x_remove(struct snd_soc_codec *codec) return 0; } -static struct regmap *pm860x_get_regmap(struct device *dev) -{ - struct pm860x_priv *pm860x = dev_get_drvdata(dev); - - return pm860x->regmap; -} - static const struct snd_soc_codec_driver soc_codec_dev_pm860x = { .probe = pm860x_probe, .remove = pm860x_remove, .set_bias_level = pm860x_set_bias_level, - .get_regmap = pm860x_get_regmap, .component_driver = { .controls = pm860x_snd_controls, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a42ddbc93f3d..f3c7758cc491 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM179X_I2C if I2C select SND_SOC_PCM179X_SPI if SPI_MASTER + select SND_SOC_PCM186X_I2C if I2C + select SND_SOC_PCM186X_SPI if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER @@ -133,7 +135,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC - select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER @@ -661,6 +662,21 @@ config SND_SOC_PCM179X_SPI Enable support for Texas Instruments PCM179x CODEC. Select this if your PCM179x is connected via an SPI bus. +config SND_SOC_PCM186X + tristate + +config SND_SOC_PCM186X_I2C + tristate "Texas Instruments PCM186x CODECs - I2C" + depends on I2C + select SND_SOC_PCM186X + select REGMAP_I2C + +config SND_SOC_PCM186X_SPI + tristate "Texas Instruments PCM186x CODECs - SPI" + depends on SPI_MASTER + select SND_SOC_PCM186X + select REGMAP_SPI + config SND_SOC_PCM3008 tristate @@ -818,9 +834,6 @@ config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO -config SND_SOC_SN95031 - tristate - config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0001069ce2a7..d930d067f602 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -105,6 +105,9 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm179x-codec-objs := pcm179x.o snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o snd-soc-pcm179x-spi-objs := pcm179x-spi.o +snd-soc-pcm186x-objs := pcm186x.o +snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o +snd-soc-pcm186x-spi-objs := pcm186x-spi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o @@ -140,7 +143,6 @@ snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o -snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2518-objs := ssm2518.o @@ -345,6 +347,9 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o obj-$(CONFIG_SND_SOC_PCM179X_SPI) += snd-soc-pcm179x-spi.o +obj-$(CONFIG_SND_SOC_PCM186X) += snd-soc-pcm186x.o +obj-$(CONFIG_SND_SOC_PCM186X_I2C) += snd-soc-pcm186x-i2c.o +obj-$(CONFIG_SND_SOC_PCM186X_SPI) += snd-soc-pcm186x-spi.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 6ed2cc374768..3bf93652bb31 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -121,17 +121,19 @@ static struct snd_soc_dai_driver cq93vc_dai = { .ops = &cq93vc_dai_ops, }; -static struct regmap *cq93vc_get_regmap(struct device *dev) +static int cq93vc_probe(struct snd_soc_component *component) { - struct davinci_vc *davinci_vc = dev->platform_data; + struct davinci_vc *davinci_vc = component->dev->platform_data; - return davinci_vc->regmap; + snd_soc_component_init_regmap(component, davinci_vc->regmap); + + return 0; } static const struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .set_bias_level = cq93vc_set_bias_level, - .get_regmap = cq93vc_get_regmap, .component_driver = { + .probe = cq93vc_probe, .controls = cq93vc_snd_controls, .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 7e9806206648..bc3a72e4c4ed 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -355,13 +355,9 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, unsigned int devid = 0; unsigned int reg; - - cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l32_private), - GFP_KERNEL); - if (!cs35l32) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + cs35l32 = devm_kzalloc(&i2c_client->dev, sizeof(*cs35l32), GFP_KERNEL); + if (!cs35l32) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs35l32); @@ -375,13 +371,11 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs35l32->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs35l32_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&i2c_client->dev, "could not allocate pdata\n"); + pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) return -ENOMEM; - } + if (i2c_client->dev.of_node) { ret = cs35l32_handle_of_data(i2c_client, &cs35l32->pdata); diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index 1e05026bedca..0600d5264c4c 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1004,13 +1004,9 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client, unsigned int devid = 0; unsigned int reg; - cs35l34 = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs35l34_private), - GFP_KERNEL); - if (!cs35l34) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + cs35l34 = devm_kzalloc(&i2c_client->dev, sizeof(*cs35l34), GFP_KERNEL); + if (!cs35l34) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs35l34); cs35l34->regmap = devm_regmap_init_i2c(i2c_client, &cs35l34_regmap); @@ -1044,14 +1040,11 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs35l34->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs35l34_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&i2c_client->dev, - "could not allocate pdata\n"); + pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) return -ENOMEM; - } + if (i2c_client->dev.of_node) { ret = cs35l34_handle_of_data(i2c_client, pdata); if (ret != 0) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0d9c4a57301b..9731e5dff291 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1100,8 +1100,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, unsigned int reg; u32 val32; - cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l52_private), - GFP_KERNEL); + cs42l52 = devm_kzalloc(&i2c_client->dev, sizeof(*cs42l52), GFP_KERNEL); if (cs42l52 == NULL) return -ENOMEM; cs42l52->dev = &i2c_client->dev; @@ -1115,13 +1114,11 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs42l52->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs42l52_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&i2c_client->dev, "could not allocate pdata\n"); + pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) return -ENOMEM; - } + if (i2c_client->dev.of_node) { if (of_property_read_bool(i2c_client->dev.of_node, "cirrus,mica-differential-cfg")) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index cb6ca85f1536..fd7b8d32c2b2 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1190,9 +1190,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, unsigned int alpha_rev, metal_rev; unsigned int reg; - cs42l56 = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs42l56_private), - GFP_KERNEL); + cs42l56 = devm_kzalloc(&i2c_client->dev, sizeof(*cs42l56), GFP_KERNEL); if (cs42l56 == NULL) return -ENOMEM; cs42l56->dev = &i2c_client->dev; @@ -1207,14 +1205,11 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs42l56->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs42l56_platform_data), + pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - dev_err(&i2c_client->dev, - "could not allocate pdata\n"); + if (!pdata) return -ENOMEM; - } + if (i2c_client->dev.of_node) { ret = cs42l56_handle_of_data(i2c_client, &cs42l56->pdata); diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3df2c473ab88..aebaa97490b6 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1289,8 +1289,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, unsigned int reg; u32 val32; - cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), - GFP_KERNEL); + cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(*cs42l73), GFP_KERNEL); if (!cs42l73) return -ENOMEM; @@ -1304,13 +1303,11 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, if (pdata) { cs42l73->pdata = *pdata; } else { - pdata = devm_kzalloc(&i2c_client->dev, - sizeof(struct cs42l73_platform_data), - GFP_KERNEL); - if (!pdata) { - dev_err(&i2c_client->dev, "could not allocate pdata\n"); + pdata = devm_kzalloc(&i2c_client->dev, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) return -ENOMEM; - } + if (i2c_client->dev.of_node) { if (of_property_read_u32(i2c_client->dev.of_node, "chgfreq", &val32) >= 0) @@ -1358,7 +1355,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, ret = regmap_read(cs42l73->regmap, CS42L73_REVID, ®); if (ret < 0) { dev_err(&i2c_client->dev, "Get Revision ID failed\n"); - return ret;; + return ret; } dev_info(&i2c_client->dev, diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 94c0209977d0..be2750680838 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1120,9 +1120,11 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct snd_soc_component *component = snd_soc_dapm_to_component(dapm); struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int ret; - priv->core.arizona->dapm = dapm; + arizona->dapm = dapm; + snd_soc_codec_init_regmap(codec, arizona->regmap); ret = arizona_init_spk(codec); if (ret < 0) @@ -1175,17 +1177,9 @@ static unsigned int cs47l24_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_4L, }; -static struct regmap *cs47l24_get_regmap(struct device *dev) -{ - struct cs47l24_priv *priv = dev_get_drvdata(dev); - - return priv->core.arizona->regmap; -} - static const struct snd_soc_codec_driver soc_codec_dev_cs47l24 = { .probe = cs47l24_codec_probe, .remove = cs47l24_codec_remove, - .get_regmap = cs47l24_get_regmap, .idle_bias_off = true, diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 46b1fbb66eba..95bb10ba80dc 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -26,8 +26,9 @@ struct cx20442_priv { - void *control_data; + struct tty_struct *tty; struct regulator *por; + u8 reg_cache; }; #define CX20442_PM 0x0 @@ -89,14 +90,14 @@ static const struct snd_soc_dapm_route cx20442_audio_map[] = { }; static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { - u8 *reg_cache = codec->reg_cache; + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); - if (reg >= codec->driver->reg_cache_size) + if (reg >= 1) return -EINVAL; - return reg_cache[reg]; + return cx20442->reg_cache; } enum v253_vls { @@ -156,20 +157,19 @@ static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); - u8 *reg_cache = codec->reg_cache; int vls, vsp, old, len; char buf[18]; - if (reg >= codec->driver->reg_cache_size) + if (reg >= 1) return -EINVAL; - /* hw_write and control_data pointers required for talking to the modem + /* tty and write pointers required for talking to the modem * are expected to be set by the line discipline initialization code */ - if (!codec->hw_write || !cx20442->control_data) + if (!cx20442->tty || !cx20442->tty->ops->write) return -EIO; - old = reg_cache[reg]; - reg_cache[reg] = value; + old = cx20442->reg_cache; + cx20442->reg_cache = value; vls = cx20442_pm_to_v253_vls(value); if (vls < 0) @@ -194,13 +194,12 @@ static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, return -ENOMEM; dev_dbg(codec->dev, "%s: %s\n", __func__, buf); - if (codec->hw_write(cx20442->control_data, buf, len) != len) + if (cx20442->tty->ops->write(cx20442->tty, buf, len) != len) return -EIO; return 0; } - /* * Line discpline related code * @@ -252,8 +251,7 @@ static void v253_close(struct tty_struct *tty) cx20442 = snd_soc_codec_get_drvdata(codec); /* Prevent the codec driver from further accessing the modem */ - codec->hw_write = NULL; - cx20442->control_data = NULL; + cx20442->tty = NULL; codec->component.card->pop_time = 0; } @@ -276,12 +274,11 @@ static void v253_receive(struct tty_struct *tty, cx20442 = snd_soc_codec_get_drvdata(codec); - if (!cx20442->control_data) { + if (!cx20442->tty) { /* First modem response, complete setup procedure */ /* Set up codec driver access to modem controls */ - cx20442->control_data = tty; - codec->hw_write = (hw_write_t)tty->ops->write; + cx20442->tty = tty; codec->component.card->pop_time = 1; } } @@ -367,10 +364,9 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442->por = regulator_get(codec->dev, "POR"); if (IS_ERR(cx20442->por)) dev_warn(codec->dev, "failed to get the regulator"); - cx20442->control_data = NULL; + cx20442->tty = NULL; snd_soc_codec_set_drvdata(codec, cx20442); - codec->hw_write = NULL; codec->component.card->pop_time = 0; return 0; @@ -381,8 +377,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) { struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); - if (cx20442->control_data) { - struct tty_struct *tty = cx20442->control_data; + if (cx20442->tty) { + struct tty_struct *tty = cx20442->tty; tty_hangup(tty); } @@ -396,17 +392,13 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) return 0; } -static const u8 cx20442_reg; - static const struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, .set_bias_level = cx20442_set_bias_level, - .reg_cache_default = &cx20442_reg, - .reg_cache_size = 1, - .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, .write = cx20442_write, + .component_driver = { .dapm_widgets = cx20442_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cx20442_dapm_widgets), diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 41d9b1da27c2..b2b4e90fc02a 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1654,10 +1654,8 @@ static struct da7213_platform_data u32 fw_val32; pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - dev_warn(codec->dev, "Failed to allocate memory for pdata\n"); + if (!pdata) return NULL; - } if (device_property_read_u32(dev, "dlg,micbias1-lvl", &fw_val32) >= 0) pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, fw_val32); @@ -1855,8 +1853,7 @@ static int da7213_i2c_probe(struct i2c_client *i2c, struct da7213_priv *da7213; int ret; - da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv), - GFP_KERNEL); + da7213 = devm_kzalloc(&i2c->dev, sizeof(*da7213), GFP_KERNEL); if (!da7213) return -ENOMEM; diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index b2d42ec1dcd9..96c644a15b11 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2455,10 +2455,8 @@ static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) u32 of_val32; pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - dev_warn(codec->dev, "Failed to allocate memory for pdata\n"); + if (!pdata) return NULL; - } if (of_property_read_u32(np, "dlg,micbias1-lvl-millivolt", &of_val32) >= 0) pdata->micbias1_lvl = da7218_of_micbias_lvl(codec, of_val32); @@ -2520,15 +2518,13 @@ static struct da7218_pdata *da7218_of_to_pdata(struct snd_soc_codec *codec) } if (da7218->dev_id == DA7218_DEV_ID) { - hpldet_np = of_find_node_by_name(np, "da7218_hpldet"); + hpldet_np = of_get_child_by_name(np, "da7218_hpldet"); if (!hpldet_np) return pdata; hpldet_pdata = devm_kzalloc(codec->dev, sizeof(*hpldet_pdata), GFP_KERNEL); if (!hpldet_pdata) { - dev_warn(codec->dev, - "Failed to allocate memory for hpldet pdata\n"); of_node_put(hpldet_np); return pdata; } @@ -3273,8 +3269,7 @@ static int da7218_i2c_probe(struct i2c_client *i2c, struct da7218_priv *da7218; int ret; - da7218 = devm_kzalloc(&i2c->dev, sizeof(struct da7218_priv), - GFP_KERNEL); + da7218 = devm_kzalloc(&i2c->dev, sizeof(*da7218), GFP_KERNEL); if (!da7218) return -ENOMEM; diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index b88a1ee66f80..c88f974ebe3e 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -107,8 +107,30 @@ static const struct snd_soc_codec_driver soc_dmic = { static int dmic_dev_probe(struct platform_device *pdev) { + int err; + u32 chans; + struct snd_soc_dai_driver *dai_drv = &dmic_dai; + + if (pdev->dev.of_node) { + err = of_property_read_u32(pdev->dev.of_node, "num-channels", &chans); + if (err && (err != -ENOENT)) + return err; + + if (!err) { + if (chans < 1 || chans > 8) + return -EINVAL; + + dai_drv = devm_kzalloc(&pdev->dev, sizeof(*dai_drv), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + + memcpy(dai_drv, &dmic_dai, sizeof(*dai_drv)); + dai_drv->capture.channels_max = chans; + } + } + return snd_soc_register_codec(&pdev->dev, - &soc_dmic, &dmic_dai, 1); + &soc_dmic, dai_drv, 1); } static int dmic_dev_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 5f3c42c4f74a..066ea2f4ce7b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -267,7 +267,7 @@ #define MSM8916_WCD_ANALOG_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index a10a724eb448..13354d6304a8 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -194,7 +194,7 @@ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_48000) #define MSM8916_WCD_DIGITAL_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) struct msm8916_wcd_digital_priv { struct clk *ahbclk, *mclk; @@ -645,7 +645,7 @@ static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, RX_I2S_CTL_RX_I2S_MODE_MASK, RX_I2S_CTL_RX_I2S_MODE_16); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: snd_soc_update_bits(dai->codec, LPASS_CDC_CLK_TX_I2S_CTL, TX_I2S_CTL_TX_I2S_MODE_MASK, TX_I2S_CTL_TX_I2S_MODE_32); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 714ce17da717..e853a6dfd33b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -905,6 +905,7 @@ static int nau8825_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + msleep(125); regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); break; diff --git a/sound/soc/codecs/pcm186x-i2c.c b/sound/soc/codecs/pcm186x-i2c.c new file mode 100644 index 000000000000..543621232d60 --- /dev/null +++ b/sound/soc/codecs/pcm186x-i2c.c @@ -0,0 +1,69 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Texas Instruments PCM186x Universal Audio ADC - I2C + * + * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Andreas Dannenberg <dannenberg@ti.com> + * Andrew F. Davis <afd@ti.com> + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "pcm186x.h" + +static const struct of_device_id pcm186x_of_match[] = { + { .compatible = "ti,pcm1862", .data = (void *)PCM1862 }, + { .compatible = "ti,pcm1863", .data = (void *)PCM1863 }, + { .compatible = "ti,pcm1864", .data = (void *)PCM1864 }, + { .compatible = "ti,pcm1865", .data = (void *)PCM1865 }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm186x_of_match); + +static int pcm186x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + const enum pcm186x_type type = (enum pcm186x_type)id->driver_data; + int irq = i2c->irq; + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm186x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm186x_probe(&i2c->dev, type, irq, regmap); +} + +static int pcm186x_i2c_remove(struct i2c_client *i2c) +{ + pcm186x_remove(&i2c->dev); + + return 0; +} + +static const struct i2c_device_id pcm186x_i2c_id[] = { + { "pcm1862", PCM1862 }, + { "pcm1863", PCM1863 }, + { "pcm1864", PCM1864 }, + { "pcm1865", PCM1865 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm186x_i2c_id); + +static struct i2c_driver pcm186x_i2c_driver = { + .probe = pcm186x_i2c_probe, + .remove = pcm186x_i2c_remove, + .id_table = pcm186x_i2c_id, + .driver = { + .name = "pcm186x", + .of_match_table = pcm186x_of_match, + }, +}; +module_i2c_driver(pcm186x_i2c_driver); + +MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>"); +MODULE_AUTHOR("Andrew F. Davis <afd@ti.com>"); +MODULE_DESCRIPTION("PCM186x Universal Audio ADC I2C Interface Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm186x-spi.c b/sound/soc/codecs/pcm186x-spi.c new file mode 100644 index 000000000000..2366f8e4d4d4 --- /dev/null +++ b/sound/soc/codecs/pcm186x-spi.c @@ -0,0 +1,69 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Texas Instruments PCM186x Universal Audio ADC - SPI + * + * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Andreas Dannenberg <dannenberg@ti.com> + * Andrew F. Davis <afd@ti.com> + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "pcm186x.h" + +static const struct of_device_id pcm186x_of_match[] = { + { .compatible = "ti,pcm1862", .data = (void *)PCM1862 }, + { .compatible = "ti,pcm1863", .data = (void *)PCM1863 }, + { .compatible = "ti,pcm1864", .data = (void *)PCM1864 }, + { .compatible = "ti,pcm1865", .data = (void *)PCM1865 }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm186x_of_match); + +static int pcm186x_spi_probe(struct spi_device *spi) +{ + const enum pcm186x_type type = + (enum pcm186x_type)spi_get_device_id(spi)->driver_data; + int irq = spi->irq; + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &pcm186x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm186x_probe(&spi->dev, type, irq, regmap); +} + +static int pcm186x_spi_remove(struct spi_device *spi) +{ + pcm186x_remove(&spi->dev); + + return 0; +} + +static const struct spi_device_id pcm186x_spi_id[] = { + { "pcm1862", PCM1862 }, + { "pcm1863", PCM1863 }, + { "pcm1864", PCM1864 }, + { "pcm1865", PCM1865 }, + { } +}; +MODULE_DEVICE_TABLE(spi, pcm186x_spi_id); + +static struct spi_driver pcm186x_spi_driver = { + .probe = pcm186x_spi_probe, + .remove = pcm186x_spi_remove, + .id_table = pcm186x_spi_id, + .driver = { + .name = "pcm186x", + .of_match_table = pcm186x_of_match, + }, +}; +module_spi_driver(pcm186x_spi_driver); + +MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>"); +MODULE_AUTHOR("Andrew F. Davis <afd@ti.com>"); +MODULE_DESCRIPTION("PCM186x Universal Audio ADC SPI Interface Driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c new file mode 100644 index 000000000000..cdb51427facc --- /dev/null +++ b/sound/soc/codecs/pcm186x.c @@ -0,0 +1,719 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Texas Instruments PCM186x Universal Audio ADC + * + * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Andreas Dannenberg <dannenberg@ti.com> + * Andrew F. Davis <afd@ti.com> + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "pcm186x.h" + +static const char * const pcm186x_supply_names[] = { + "avdd", /* Analog power supply. Connect to 3.3-V supply. */ + "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ + "iovdd", /* I/O power supply. Connect to 3.3-V or 1.8-V. */ +}; +#define PCM186x_NUM_SUPPLIES ARRAY_SIZE(pcm186x_supply_names) + +struct pcm186x_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[PCM186x_NUM_SUPPLIES]; + unsigned int sysclk; + unsigned int tdm_offset; + bool is_tdm_mode; + bool is_master_mode; +}; + +static const DECLARE_TLV_DB_SCALE(pcm186x_pga_tlv, -1200, 4000, 50); + +static const struct snd_kcontrol_new pcm1863_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", PCM186X_PGA_VAL_CH1_L, + PCM186X_PGA_VAL_CH1_R, 0, -24, 80, 7, 0, + pcm186x_pga_tlv), +}; + +static const struct snd_kcontrol_new pcm1865_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", PCM186X_PGA_VAL_CH1_L, + PCM186X_PGA_VAL_CH1_R, 0, -24, 80, 7, 0, + pcm186x_pga_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", PCM186X_PGA_VAL_CH2_L, + PCM186X_PGA_VAL_CH2_R, 0, -24, 80, 7, 0, + pcm186x_pga_tlv), +}; + +static const unsigned int pcm186x_adc_input_channel_sel_value[] = { + 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, + 0x08, 0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, + 0x10, 0x20, 0x30 +}; + +static const char * const pcm186x_adcl_input_channel_sel_text[] = { + "No Select", + "VINL1[SE]", /* Default for ADC1L */ + "VINL2[SE]", /* Default for ADC2L */ + "VINL2[SE] + VINL1[SE]", + "VINL3[SE]", + "VINL3[SE] + VINL1[SE]", + "VINL3[SE] + VINL2[SE]", + "VINL3[SE] + VINL2[SE] + VINL1[SE]", + "VINL4[SE]", + "VINL4[SE] + VINL1[SE]", + "VINL4[SE] + VINL2[SE]", + "VINL4[SE] + VINL2[SE] + VINL1[SE]", + "VINL4[SE] + VINL3[SE]", + "VINL4[SE] + VINL3[SE] + VINL1[SE]", + "VINL4[SE] + VINL3[SE] + VINL2[SE]", + "VINL4[SE] + VINL3[SE] + VINL2[SE] + VINL1[SE]", + "{VIN1P, VIN1M}[DIFF]", + "{VIN4P, VIN4M}[DIFF]", + "{VIN1P, VIN1M}[DIFF] + {VIN4P, VIN4M}[DIFF]" +}; + +static const char * const pcm186x_adcr_input_channel_sel_text[] = { + "No Select", + "VINR1[SE]", /* Default for ADC1R */ + "VINR2[SE]", /* Default for ADC2R */ + "VINR2[SE] + VINR1[SE]", + "VINR3[SE]", + "VINR3[SE] + VINR1[SE]", + "VINR3[SE] + VINR2[SE]", + "VINR3[SE] + VINR2[SE] + VINR1[SE]", + "VINR4[SE]", + "VINR4[SE] + VINR1[SE]", + "VINR4[SE] + VINR2[SE]", + "VINR4[SE] + VINR2[SE] + VINR1[SE]", + "VINR4[SE] + VINR3[SE]", + "VINR4[SE] + VINR3[SE] + VINR1[SE]", + "VINR4[SE] + VINR3[SE] + VINR2[SE]", + "VINR4[SE] + VINR3[SE] + VINR2[SE] + VINR1[SE]", + "{VIN2P, VIN2M}[DIFF]", + "{VIN3P, VIN3M}[DIFF]", + "{VIN2P, VIN2M}[DIFF] + {VIN3P, VIN3M}[DIFF]" +}; + +static const struct soc_enum pcm186x_adc_input_channel_sel[] = { + SOC_VALUE_ENUM_SINGLE(PCM186X_ADC1_INPUT_SEL_L, 0, + PCM186X_ADC_INPUT_SEL_MASK, + ARRAY_SIZE(pcm186x_adcl_input_channel_sel_text), + pcm186x_adcl_input_channel_sel_text, + pcm186x_adc_input_channel_sel_value), + SOC_VALUE_ENUM_SINGLE(PCM186X_ADC1_INPUT_SEL_R, 0, + PCM186X_ADC_INPUT_SEL_MASK, + ARRAY_SIZE(pcm186x_adcr_input_channel_sel_text), + pcm186x_adcr_input_channel_sel_text, + pcm186x_adc_input_channel_sel_value), + SOC_VALUE_ENUM_SINGLE(PCM186X_ADC2_INPUT_SEL_L, 0, + PCM186X_ADC_INPUT_SEL_MASK, + ARRAY_SIZE(pcm186x_adcl_input_channel_sel_text), + pcm186x_adcl_input_channel_sel_text, + pcm186x_adc_input_channel_sel_value), + SOC_VALUE_ENUM_SINGLE(PCM186X_ADC2_INPUT_SEL_R, 0, + PCM186X_ADC_INPUT_SEL_MASK, + ARRAY_SIZE(pcm186x_adcr_input_channel_sel_text), + pcm186x_adcr_input_channel_sel_text, + pcm186x_adc_input_channel_sel_value), +}; + +static const struct snd_kcontrol_new pcm186x_adc_mux_controls[] = { + SOC_DAPM_ENUM("ADC1 Left Input", pcm186x_adc_input_channel_sel[0]), + SOC_DAPM_ENUM("ADC1 Right Input", pcm186x_adc_input_channel_sel[1]), + SOC_DAPM_ENUM("ADC2 Left Input", pcm186x_adc_input_channel_sel[2]), + SOC_DAPM_ENUM("ADC2 Right Input", pcm186x_adc_input_channel_sel[3]), +}; + +static const struct snd_soc_dapm_widget pcm1863_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_INPUT("VINL3"), + SND_SOC_DAPM_INPUT("VINR3"), + SND_SOC_DAPM_INPUT("VINL4"), + SND_SOC_DAPM_INPUT("VINR4"), + + SND_SOC_DAPM_MUX("ADC Left Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[0]), + SND_SOC_DAPM_MUX("ADC Right Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[1]), + + /* + * Put the codec into SLEEP mode when not in use, allowing the + * Energysense mechanism to operate. + */ + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", PCM186X_POWER_CTRL, 1, 0), +}; + +static const struct snd_soc_dapm_widget pcm1865_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_INPUT("VINL3"), + SND_SOC_DAPM_INPUT("VINR3"), + SND_SOC_DAPM_INPUT("VINL4"), + SND_SOC_DAPM_INPUT("VINR4"), + + SND_SOC_DAPM_MUX("ADC1 Left Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[0]), + SND_SOC_DAPM_MUX("ADC1 Right Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[1]), + SND_SOC_DAPM_MUX("ADC2 Left Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[2]), + SND_SOC_DAPM_MUX("ADC2 Right Capture Source", SND_SOC_NOPM, 0, 0, + &pcm186x_adc_mux_controls[3]), + + /* + * Put the codec into SLEEP mode when not in use, allowing the + * Energysense mechanism to operate. + */ + SND_SOC_DAPM_ADC("ADC1", "HiFi Capture 1", PCM186X_POWER_CTRL, 1, 0), + SND_SOC_DAPM_ADC("ADC2", "HiFi Capture 2", PCM186X_POWER_CTRL, 1, 0), +}; + +static const struct snd_soc_dapm_route pcm1863_dapm_routes[] = { + { "ADC Left Capture Source", NULL, "VINL1" }, + { "ADC Left Capture Source", NULL, "VINR1" }, + { "ADC Left Capture Source", NULL, "VINL2" }, + { "ADC Left Capture Source", NULL, "VINR2" }, + { "ADC Left Capture Source", NULL, "VINL3" }, + { "ADC Left Capture Source", NULL, "VINR3" }, + { "ADC Left Capture Source", NULL, "VINL4" }, + { "ADC Left Capture Source", NULL, "VINR4" }, + + { "ADC", NULL, "ADC Left Capture Source" }, + + { "ADC Right Capture Source", NULL, "VINL1" }, + { "ADC Right Capture Source", NULL, "VINR1" }, + { "ADC Right Capture Source", NULL, "VINL2" }, + { "ADC Right Capture Source", NULL, "VINR2" }, + { "ADC Right Capture Source", NULL, "VINL3" }, + { "ADC Right Capture Source", NULL, "VINR3" }, + { "ADC Right Capture Source", NULL, "VINL4" }, + { "ADC Right Capture Source", NULL, "VINR4" }, + + { "ADC", NULL, "ADC Right Capture Source" }, +}; + +static const struct snd_soc_dapm_route pcm1865_dapm_routes[] = { + { "ADC1 Left Capture Source", NULL, "VINL1" }, + { "ADC1 Left Capture Source", NULL, "VINR1" }, + { "ADC1 Left Capture Source", NULL, "VINL2" }, + { "ADC1 Left Capture Source", NULL, "VINR2" }, + { "ADC1 Left Capture Source", NULL, "VINL3" }, + { "ADC1 Left Capture Source", NULL, "VINR3" }, + { "ADC1 Left Capture Source", NULL, "VINL4" }, + { "ADC1 Left Capture Source", NULL, "VINR4" }, + + { "ADC1", NULL, "ADC1 Left Capture Source" }, + + { "ADC1 Right Capture Source", NULL, "VINL1" }, + { "ADC1 Right Capture Source", NULL, "VINR1" }, + { "ADC1 Right Capture Source", NULL, "VINL2" }, + { "ADC1 Right Capture Source", NULL, "VINR2" }, + { "ADC1 Right Capture Source", NULL, "VINL3" }, + { "ADC1 Right Capture Source", NULL, "VINR3" }, + { "ADC1 Right Capture Source", NULL, "VINL4" }, + { "ADC1 Right Capture Source", NULL, "VINR4" }, + + { "ADC1", NULL, "ADC1 Right Capture Source" }, + + { "ADC2 Left Capture Source", NULL, "VINL1" }, + { "ADC2 Left Capture Source", NULL, "VINR1" }, + { "ADC2 Left Capture Source", NULL, "VINL2" }, + { "ADC2 Left Capture Source", NULL, "VINR2" }, + { "ADC2 Left Capture Source", NULL, "VINL3" }, + { "ADC2 Left Capture Source", NULL, "VINR3" }, + { "ADC2 Left Capture Source", NULL, "VINL4" }, + { "ADC2 Left Capture Source", NULL, "VINR4" }, + + { "ADC2", NULL, "ADC2 Left Capture Source" }, + + { "ADC2 Right Capture Source", NULL, "VINL1" }, + { "ADC2 Right Capture Source", NULL, "VINR1" }, + { "ADC2 Right Capture Source", NULL, "VINL2" }, + { "ADC2 Right Capture Source", NULL, "VINR2" }, + { "ADC2 Right Capture Source", NULL, "VINL3" }, + { "ADC2 Right Capture Source", NULL, "VINR3" }, + { "ADC2 Right Capture Source", NULL, "VINL4" }, + { "ADC2 Right Capture Source", NULL, "VINR4" }, + + { "ADC2", NULL, "ADC2 Right Capture Source" }, +}; + +static int pcm186x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int format = params_format(params); + unsigned int width = params_width(params); + unsigned int channels = params_channels(params); + unsigned int div_lrck; + unsigned int div_bck; + u8 tdm_tx_sel = 0; + u8 pcm_cfg = 0; + + dev_dbg(codec->dev, "%s() rate=%u format=0x%x width=%u channels=%u\n", + __func__, rate, format, width, channels); + + switch (width) { + case 16: + pcm_cfg = PCM186X_PCM_CFG_RX_WLEN_16 << + PCM186X_PCM_CFG_RX_WLEN_SHIFT | + PCM186X_PCM_CFG_TX_WLEN_16 << + PCM186X_PCM_CFG_TX_WLEN_SHIFT; + break; + case 20: + pcm_cfg = PCM186X_PCM_CFG_RX_WLEN_20 << + PCM186X_PCM_CFG_RX_WLEN_SHIFT | + PCM186X_PCM_CFG_TX_WLEN_20 << + PCM186X_PCM_CFG_TX_WLEN_SHIFT; + break; + case 24: + pcm_cfg = PCM186X_PCM_CFG_RX_WLEN_24 << + PCM186X_PCM_CFG_RX_WLEN_SHIFT | + PCM186X_PCM_CFG_TX_WLEN_24 << + PCM186X_PCM_CFG_TX_WLEN_SHIFT; + break; + case 32: + pcm_cfg = PCM186X_PCM_CFG_RX_WLEN_32 << + PCM186X_PCM_CFG_RX_WLEN_SHIFT | + PCM186X_PCM_CFG_TX_WLEN_32 << + PCM186X_PCM_CFG_TX_WLEN_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, PCM186X_PCM_CFG, + PCM186X_PCM_CFG_RX_WLEN_MASK | + PCM186X_PCM_CFG_TX_WLEN_MASK, + pcm_cfg); + + div_lrck = width * channels; + + if (priv->is_tdm_mode) { + /* Select TDM transmission data */ + switch (channels) { + case 2: + tdm_tx_sel = PCM186X_TDM_TX_SEL_2CH; + break; + case 4: + tdm_tx_sel = PCM186X_TDM_TX_SEL_4CH; + break; + case 6: + tdm_tx_sel = PCM186X_TDM_TX_SEL_6CH; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, PCM186X_TDM_TX_SEL, + PCM186X_TDM_TX_SEL_MASK, tdm_tx_sel); + + /* In DSP/TDM mode, the LRCLK divider must be 256 */ + div_lrck = 256; + + /* Configure 1/256 duty cycle for LRCK */ + snd_soc_update_bits(codec, PCM186X_PCM_CFG, + PCM186X_PCM_CFG_TDM_LRCK_MODE, + PCM186X_PCM_CFG_TDM_LRCK_MODE); + } + + /* Only configure clock dividers in master mode. */ + if (priv->is_master_mode) { + div_bck = priv->sysclk / (div_lrck * rate); + + dev_dbg(codec->dev, + "%s() master_clk=%u div_bck=%u div_lrck=%u\n", + __func__, priv->sysclk, div_bck, div_lrck); + + snd_soc_write(codec, PCM186X_BCK_DIV, div_bck - 1); + snd_soc_write(codec, PCM186X_LRK_DIV, div_lrck - 1); + } + + return 0; +} + +static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + u8 clk_ctrl = 0; + u8 pcm_cfg = 0; + + dev_dbg(codec->dev, "%s() format=0x%x\n", __func__, format); + + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + if (!priv->sysclk) { + dev_err(codec->dev, "operating in master mode requires sysclock to be configured\n"); + return -EINVAL; + } + clk_ctrl |= PCM186X_CLK_CTRL_MST_MODE; + priv->is_master_mode = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + priv->is_master_mode = false; + break; + default: + dev_err(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* set interface polarity */ + switch (format & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + dev_err(codec->dev, "Inverted DAI clocks not supported\n"); + return -EINVAL; + } + + /* set interface format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + pcm_cfg = PCM186X_PCM_CFG_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + pcm_cfg = PCM186X_PCM_CFG_FMT_LEFTJ; + break; + case SND_SOC_DAIFMT_DSP_A: + priv->tdm_offset += 1; + /* Fall through... DSP_A uses the same basic config as DSP_B + * except we need to shift the TDM output by one BCK cycle + */ + case SND_SOC_DAIFMT_DSP_B: + priv->is_tdm_mode = true; + pcm_cfg = PCM186X_PCM_CFG_FMT_TDM; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, PCM186X_CLK_CTRL, + PCM186X_CLK_CTRL_MST_MODE, clk_ctrl); + + snd_soc_write(codec, PCM186X_TDM_TX_OFFSET, priv->tdm_offset); + + snd_soc_update_bits(codec, PCM186X_PCM_CFG, + PCM186X_PCM_CFG_FMT_MASK, pcm_cfg); + + return 0; +} + +static int pcm186x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int first_slot, last_slot, tdm_offset; + + dev_dbg(codec->dev, + "%s() tx_mask=0x%x rx_mask=0x%x slots=%d slot_width=%d\n", + __func__, tx_mask, rx_mask, slots, slot_width); + + if (!tx_mask) { + dev_err(codec->dev, "tdm tx mask must not be 0\n"); + return -EINVAL; + } + + first_slot = __ffs(tx_mask); + last_slot = __fls(tx_mask); + + if (last_slot - first_slot != hweight32(tx_mask) - 1) { + dev_err(codec->dev, "tdm tx mask must be contiguous\n"); + return -EINVAL; + } + + tdm_offset = first_slot * slot_width; + + if (tdm_offset > 255) { + dev_err(codec->dev, "tdm tx slot selection out of bounds\n"); + return -EINVAL; + } + + priv->tdm_offset = tdm_offset; + + return 0; +} + +static int pcm186x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s() clk_id=%d freq=%u dir=%d\n", + __func__, clk_id, freq, dir); + + priv->sysclk = freq; + + return 0; +} + +static const struct snd_soc_dai_ops pcm186x_dai_ops = { + .set_sysclk = pcm186x_set_dai_sysclk, + .set_tdm_slot = pcm186x_set_tdm_slot, + .set_fmt = pcm186x_set_fmt, + .hw_params = pcm186x_hw_params, +}; + +static struct snd_soc_dai_driver pcm1863_dai = { + .name = "pcm1863-aif", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM186X_RATES, + .formats = PCM186X_FORMATS, + }, + .ops = &pcm186x_dai_ops, +}; + +static struct snd_soc_dai_driver pcm1865_dai = { + .name = "pcm1865-aif", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = PCM186X_RATES, + .formats = PCM186X_FORMATS, + }, + .ops = &pcm186x_dai_ops, +}; + +static int pcm186x_power_on(struct snd_soc_codec *codec) +{ + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) + return ret; + + regcache_cache_only(priv->regmap, false); + ret = regcache_sync(priv->regmap); + if (ret) { + dev_err(codec->dev, "Failed to restore cache\n"); + regcache_cache_only(priv->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + return ret; + } + + snd_soc_update_bits(codec, PCM186X_POWER_CTRL, + PCM186X_PWR_CTRL_PWRDN, 0); + + return 0; +} + +static int pcm186x_power_off(struct snd_soc_codec *codec) +{ + struct pcm186x_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + snd_soc_update_bits(codec, PCM186X_POWER_CTRL, + PCM186X_PWR_CTRL_PWRDN, PCM186X_PWR_CTRL_PWRDN); + + regcache_cache_only(priv->regmap, true); + + ret = regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) + return ret; + + return 0; +} + +static int pcm186x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + snd_soc_codec_get_bias_level(codec), level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) + pcm186x_power_on(codec); + break; + case SND_SOC_BIAS_OFF: + pcm186x_power_off(codec); + break; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pcm1863 = { + .set_bias_level = pcm186x_set_bias_level, + + .component_driver = { + .controls = pcm1863_snd_controls, + .num_controls = ARRAY_SIZE(pcm1863_snd_controls), + .dapm_widgets = pcm1863_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1863_dapm_widgets), + .dapm_routes = pcm1863_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1863_dapm_routes), + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1865 = { + .set_bias_level = pcm186x_set_bias_level, + .suspend_bias_off = true, + + .component_driver = { + .controls = pcm1865_snd_controls, + .num_controls = ARRAY_SIZE(pcm1865_snd_controls), + .dapm_widgets = pcm1865_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1865_dapm_widgets), + .dapm_routes = pcm1865_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1865_dapm_routes), + }, +}; + +static bool pcm186x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM186X_PAGE: + case PCM186X_DEVICE_STATUS: + case PCM186X_FSAMPLE_STATUS: + case PCM186X_DIV_STATUS: + case PCM186X_CLK_STATUS: + case PCM186X_SUPPLY_STATUS: + case PCM186X_MMAP_STAT_CTRL: + case PCM186X_MMAP_ADDRESS: + return true; + } + + return false; +} + +static const struct regmap_range_cfg pcm186x_range = { + .name = "Pages", + .range_max = PCM186X_MAX_REGISTER, + .selector_reg = PCM186X_PAGE, + .selector_mask = 0xff, + .window_len = PCM186X_PAGE_LEN, +}; + +const struct regmap_config pcm186x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .volatile_reg = pcm186x_volatile, + + .ranges = &pcm186x_range, + .num_ranges = 1, + + .max_register = PCM186X_MAX_REGISTER, + + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(pcm186x_regmap); + +int pcm186x_probe(struct device *dev, enum pcm186x_type type, int irq, + struct regmap *regmap) +{ + struct pcm186x_priv *priv; + int i, ret; + + priv = devm_kzalloc(dev, sizeof(struct pcm186x_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(dev, priv); + priv->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(priv->supplies); i++) + priv->supplies[i].supply = pcm186x_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "failed enable supplies: %d\n", ret); + return ret; + } + + /* Reset device registers for a consistent power-on like state */ + ret = regmap_write(regmap, PCM186X_PAGE, PCM186X_RESET); + if (ret) { + dev_err(dev, "failed to write device: %d\n", ret); + return ret; + } + + ret = regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret) { + dev_err(dev, "failed disable supplies: %d\n", ret); + return ret; + } + + switch (type) { + case PCM1865: + case PCM1864: + ret = snd_soc_register_codec(dev, &soc_codec_dev_pcm1865, + &pcm1865_dai, 1); + break; + case PCM1863: + case PCM1862: + default: + ret = snd_soc_register_codec(dev, &soc_codec_dev_pcm1863, + &pcm1863_dai, 1); + } + if (ret) { + dev_err(dev, "failed to register CODEC: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(pcm186x_probe); + +int pcm186x_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + + return 0; +} +EXPORT_SYMBOL_GPL(pcm186x_remove); + +MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>"); +MODULE_AUTHOR("Andrew F. Davis <afd@ti.com>"); +MODULE_DESCRIPTION("PCM186x Universal Audio ADC driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm186x.h b/sound/soc/codecs/pcm186x.h new file mode 100644 index 000000000000..b630111bb3c4 --- /dev/null +++ b/sound/soc/codecs/pcm186x.h @@ -0,0 +1,220 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Texas Instruments PCM186x Universal Audio ADC + * + * Copyright (C) 2015-2017 Texas Instruments Incorporated - http://www.ti.com + * Andreas Dannenberg <dannenberg@ti.com> + * Andrew F. Davis <afd@ti.com> + */ + +#ifndef _PCM186X_H_ +#define _PCM186X_H_ + +#include <linux/pm.h> +#include <linux/regmap.h> + +enum pcm186x_type { + PCM1862, + PCM1863, + PCM1864, + PCM1865, +}; + +#define PCM186X_RATES SNDRV_PCM_RATE_8000_192000 +#define PCM186X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PCM186X_PAGE_LEN 0x0100 +#define PCM186X_PAGE_BASE(n) (PCM186X_PAGE_LEN * n) + +/* The page selection register address is the same on all pages */ +#define PCM186X_PAGE 0 + +/* Register Definitions - Page 0 */ +#define PCM186X_PGA_VAL_CH1_L (PCM186X_PAGE_BASE(0) + 1) +#define PCM186X_PGA_VAL_CH1_R (PCM186X_PAGE_BASE(0) + 2) +#define PCM186X_PGA_VAL_CH2_L (PCM186X_PAGE_BASE(0) + 3) +#define PCM186X_PGA_VAL_CH2_R (PCM186X_PAGE_BASE(0) + 4) +#define PCM186X_PGA_CTRL (PCM186X_PAGE_BASE(0) + 5) +#define PCM186X_ADC1_INPUT_SEL_L (PCM186X_PAGE_BASE(0) + 6) +#define PCM186X_ADC1_INPUT_SEL_R (PCM186X_PAGE_BASE(0) + 7) +#define PCM186X_ADC2_INPUT_SEL_L (PCM186X_PAGE_BASE(0) + 8) +#define PCM186X_ADC2_INPUT_SEL_R (PCM186X_PAGE_BASE(0) + 9) +#define PCM186X_AUXADC_INPUT_SEL (PCM186X_PAGE_BASE(0) + 10) +#define PCM186X_PCM_CFG (PCM186X_PAGE_BASE(0) + 11) +#define PCM186X_TDM_TX_SEL (PCM186X_PAGE_BASE(0) + 12) +#define PCM186X_TDM_TX_OFFSET (PCM186X_PAGE_BASE(0) + 13) +#define PCM186X_TDM_RX_OFFSET (PCM186X_PAGE_BASE(0) + 14) +#define PCM186X_DPGA_VAL_CH1_L (PCM186X_PAGE_BASE(0) + 15) +#define PCM186X_GPIO1_0_CTRL (PCM186X_PAGE_BASE(0) + 16) +#define PCM186X_GPIO3_2_CTRL (PCM186X_PAGE_BASE(0) + 17) +#define PCM186X_GPIO1_0_DIR_CTRL (PCM186X_PAGE_BASE(0) + 18) +#define PCM186X_GPIO3_2_DIR_CTRL (PCM186X_PAGE_BASE(0) + 19) +#define PCM186X_GPIO_IN_OUT (PCM186X_PAGE_BASE(0) + 20) +#define PCM186X_GPIO_PULL_CTRL (PCM186X_PAGE_BASE(0) + 21) +#define PCM186X_DPGA_VAL_CH1_R (PCM186X_PAGE_BASE(0) + 22) +#define PCM186X_DPGA_VAL_CH2_L (PCM186X_PAGE_BASE(0) + 23) +#define PCM186X_DPGA_VAL_CH2_R (PCM186X_PAGE_BASE(0) + 24) +#define PCM186X_DPGA_GAIN_CTRL (PCM186X_PAGE_BASE(0) + 25) +#define PCM186X_DPGA_MIC_CTRL (PCM186X_PAGE_BASE(0) + 26) +#define PCM186X_DIN_RESAMP_CTRL (PCM186X_PAGE_BASE(0) + 27) +#define PCM186X_CLK_CTRL (PCM186X_PAGE_BASE(0) + 32) +#define PCM186X_DSP1_CLK_DIV (PCM186X_PAGE_BASE(0) + 33) +#define PCM186X_DSP2_CLK_DIV (PCM186X_PAGE_BASE(0) + 34) +#define PCM186X_ADC_CLK_DIV (PCM186X_PAGE_BASE(0) + 35) +#define PCM186X_PLL_SCK_DIV (PCM186X_PAGE_BASE(0) + 37) +#define PCM186X_BCK_DIV (PCM186X_PAGE_BASE(0) + 38) +#define PCM186X_LRK_DIV (PCM186X_PAGE_BASE(0) + 39) +#define PCM186X_PLL_CTRL (PCM186X_PAGE_BASE(0) + 40) +#define PCM186X_PLL_P_DIV (PCM186X_PAGE_BASE(0) + 41) +#define PCM186X_PLL_R_DIV (PCM186X_PAGE_BASE(0) + 42) +#define PCM186X_PLL_J_DIV (PCM186X_PAGE_BASE(0) + 43) +#define PCM186X_PLL_D_DIV_LSB (PCM186X_PAGE_BASE(0) + 44) +#define PCM186X_PLL_D_DIV_MSB (PCM186X_PAGE_BASE(0) + 45) +#define PCM186X_SIGDET_MODE (PCM186X_PAGE_BASE(0) + 48) +#define PCM186X_SIGDET_MASK (PCM186X_PAGE_BASE(0) + 49) +#define PCM186X_SIGDET_STAT (PCM186X_PAGE_BASE(0) + 50) +#define PCM186X_SIGDET_LOSS_TIME (PCM186X_PAGE_BASE(0) + 52) +#define PCM186X_SIGDET_SCAN_TIME (PCM186X_PAGE_BASE(0) + 53) +#define PCM186X_SIGDET_INT_INTVL (PCM186X_PAGE_BASE(0) + 54) +#define PCM186X_SIGDET_DC_REF_CH1_L (PCM186X_PAGE_BASE(0) + 64) +#define PCM186X_SIGDET_DC_DIFF_CH1_L (PCM186X_PAGE_BASE(0) + 65) +#define PCM186X_SIGDET_DC_LEV_CH1_L (PCM186X_PAGE_BASE(0) + 66) +#define PCM186X_SIGDET_DC_REF_CH1_R (PCM186X_PAGE_BASE(0) + 67) +#define PCM186X_SIGDET_DC_DIFF_CH1_R (PCM186X_PAGE_BASE(0) + 68) +#define PCM186X_SIGDET_DC_LEV_CH1_R (PCM186X_PAGE_BASE(0) + 69) +#define PCM186X_SIGDET_DC_REF_CH2_L (PCM186X_PAGE_BASE(0) + 70) +#define PCM186X_SIGDET_DC_DIFF_CH2_L (PCM186X_PAGE_BASE(0) + 71) +#define PCM186X_SIGDET_DC_LEV_CH2_L (PCM186X_PAGE_BASE(0) + 72) +#define PCM186X_SIGDET_DC_REF_CH2_R (PCM186X_PAGE_BASE(0) + 73) +#define PCM186X_SIGDET_DC_DIFF_CH2_R (PCM186X_PAGE_BASE(0) + 74) +#define PCM186X_SIGDET_DC_LEV_CH2_R (PCM186X_PAGE_BASE(0) + 75) +#define PCM186X_SIGDET_DC_REF_CH3_L (PCM186X_PAGE_BASE(0) + 76) +#define PCM186X_SIGDET_DC_DIFF_CH3_L (PCM186X_PAGE_BASE(0) + 77) +#define PCM186X_SIGDET_DC_LEV_CH3_L (PCM186X_PAGE_BASE(0) + 78) +#define PCM186X_SIGDET_DC_REF_CH3_R (PCM186X_PAGE_BASE(0) + 79) +#define PCM186X_SIGDET_DC_DIFF_CH3_R (PCM186X_PAGE_BASE(0) + 80) +#define PCM186X_SIGDET_DC_LEV_CH3_R (PCM186X_PAGE_BASE(0) + 81) +#define PCM186X_SIGDET_DC_REF_CH4_L (PCM186X_PAGE_BASE(0) + 82) +#define PCM186X_SIGDET_DC_DIFF_CH4_L (PCM186X_PAGE_BASE(0) + 83) +#define PCM186X_SIGDET_DC_LEV_CH4_L (PCM186X_PAGE_BASE(0) + 84) +#define PCM186X_SIGDET_DC_REF_CH4_R (PCM186X_PAGE_BASE(0) + 85) +#define PCM186X_SIGDET_DC_DIFF_CH4_R (PCM186X_PAGE_BASE(0) + 86) +#define PCM186X_SIGDET_DC_LEV_CH4_R (PCM186X_PAGE_BASE(0) + 87) +#define PCM186X_AUXADC_DATA_CTRL (PCM186X_PAGE_BASE(0) + 88) +#define PCM186X_AUXADC_DATA_LSB (PCM186X_PAGE_BASE(0) + 89) +#define PCM186X_AUXADC_DATA_MSB (PCM186X_PAGE_BASE(0) + 90) +#define PCM186X_INT_ENABLE (PCM186X_PAGE_BASE(0) + 96) +#define PCM186X_INT_FLAG (PCM186X_PAGE_BASE(0) + 97) +#define PCM186X_INT_POL_WIDTH (PCM186X_PAGE_BASE(0) + 98) +#define PCM186X_POWER_CTRL (PCM186X_PAGE_BASE(0) + 112) +#define PCM186X_FILTER_MUTE_CTRL (PCM186X_PAGE_BASE(0) + 113) +#define PCM186X_DEVICE_STATUS (PCM186X_PAGE_BASE(0) + 114) +#define PCM186X_FSAMPLE_STATUS (PCM186X_PAGE_BASE(0) + 115) +#define PCM186X_DIV_STATUS (PCM186X_PAGE_BASE(0) + 116) +#define PCM186X_CLK_STATUS (PCM186X_PAGE_BASE(0) + 117) +#define PCM186X_SUPPLY_STATUS (PCM186X_PAGE_BASE(0) + 120) + +/* Register Definitions - Page 1 */ +#define PCM186X_MMAP_STAT_CTRL (PCM186X_PAGE_BASE(1) + 1) +#define PCM186X_MMAP_ADDRESS (PCM186X_PAGE_BASE(1) + 2) +#define PCM186X_MEM_WDATA0 (PCM186X_PAGE_BASE(1) + 4) +#define PCM186X_MEM_WDATA1 (PCM186X_PAGE_BASE(1) + 5) +#define PCM186X_MEM_WDATA2 (PCM186X_PAGE_BASE(1) + 6) +#define PCM186X_MEM_WDATA3 (PCM186X_PAGE_BASE(1) + 7) +#define PCM186X_MEM_RDATA0 (PCM186X_PAGE_BASE(1) + 8) +#define PCM186X_MEM_RDATA1 (PCM186X_PAGE_BASE(1) + 9) +#define PCM186X_MEM_RDATA2 (PCM186X_PAGE_BASE(1) + 10) +#define PCM186X_MEM_RDATA3 (PCM186X_PAGE_BASE(1) + 11) + +/* Register Definitions - Page 3 */ +#define PCM186X_OSC_PWR_DOWN_CTRL (PCM186X_PAGE_BASE(3) + 18) +#define PCM186X_MIC_BIAS_CTRL (PCM186X_PAGE_BASE(3) + 21) + +/* Register Definitions - Page 253 */ +#define PCM186X_CURR_TRIM_CTRL (PCM186X_PAGE_BASE(253) + 20) + +#define PCM186X_MAX_REGISTER PCM186X_CURR_TRIM_CTRL + +/* PCM186X_PAGE */ +#define PCM186X_RESET 0xff + +/* PCM186X_ADCX_INPUT_SEL_X */ +#define PCM186X_ADC_INPUT_SEL_POL BIT(7) +#define PCM186X_ADC_INPUT_SEL_MASK GENMASK(5, 0) + +/* PCM186X_PCM_CFG */ +#define PCM186X_PCM_CFG_RX_WLEN_MASK GENMASK(7, 6) +#define PCM186X_PCM_CFG_RX_WLEN_SHIFT 6 +#define PCM186X_PCM_CFG_RX_WLEN_32 0x00 +#define PCM186X_PCM_CFG_RX_WLEN_24 0x01 +#define PCM186X_PCM_CFG_RX_WLEN_20 0x02 +#define PCM186X_PCM_CFG_RX_WLEN_16 0x03 +#define PCM186X_PCM_CFG_TDM_LRCK_MODE BIT(4) +#define PCM186X_PCM_CFG_TX_WLEN_MASK GENMASK(3, 2) +#define PCM186X_PCM_CFG_TX_WLEN_SHIFT 2 +#define PCM186X_PCM_CFG_TX_WLEN_32 0x00 +#define PCM186X_PCM_CFG_TX_WLEN_24 0x01 +#define PCM186X_PCM_CFG_TX_WLEN_20 0x02 +#define PCM186X_PCM_CFG_TX_WLEN_16 0x03 +#define PCM186X_PCM_CFG_FMT_MASK GENMASK(1, 0) +#define PCM186X_PCM_CFG_FMT_SHIFT 0 +#define PCM186X_PCM_CFG_FMT_I2S 0x00 +#define PCM186X_PCM_CFG_FMT_LEFTJ 0x01 +#define PCM186X_PCM_CFG_FMT_RIGHTJ 0x02 +#define PCM186X_PCM_CFG_FMT_TDM 0x03 + +/* PCM186X_TDM_TX_SEL */ +#define PCM186X_TDM_TX_SEL_2CH 0x00 +#define PCM186X_TDM_TX_SEL_4CH 0x01 +#define PCM186X_TDM_TX_SEL_6CH 0x02 +#define PCM186X_TDM_TX_SEL_MASK 0x03 + +/* PCM186X_CLK_CTRL */ +#define PCM186X_CLK_CTRL_SCK_XI_SEL1 BIT(7) +#define PCM186X_CLK_CTRL_SCK_XI_SEL0 BIT(6) +#define PCM186X_CLK_CTRL_SCK_SRC_PLL BIT(5) +#define PCM186X_CLK_CTRL_MST_MODE BIT(4) +#define PCM186X_CLK_CTRL_ADC_SRC_PLL BIT(3) +#define PCM186X_CLK_CTRL_DSP2_SRC_PLL BIT(2) +#define PCM186X_CLK_CTRL_DSP1_SRC_PLL BIT(1) +#define PCM186X_CLK_CTRL_CLKDET_EN BIT(0) + +/* PCM186X_PLL_CTRL */ +#define PCM186X_PLL_CTRL_LOCK BIT(4) +#define PCM186X_PLL_CTRL_REF_SEL BIT(1) +#define PCM186X_PLL_CTRL_EN BIT(0) + +/* PCM186X_POWER_CTRL */ +#define PCM186X_PWR_CTRL_PWRDN BIT(2) +#define PCM186X_PWR_CTRL_SLEEP BIT(1) +#define PCM186X_PWR_CTRL_STBY BIT(0) + +/* PCM186X_CLK_STATUS */ +#define PCM186X_CLK_STATUS_LRCKHLT BIT(6) +#define PCM186X_CLK_STATUS_BCKHLT BIT(5) +#define PCM186X_CLK_STATUS_SCKHLT BIT(4) +#define PCM186X_CLK_STATUS_LRCKERR BIT(2) +#define PCM186X_CLK_STATUS_BCKERR BIT(1) +#define PCM186X_CLK_STATUS_SCKERR BIT(0) + +/* PCM186X_SUPPLY_STATUS */ +#define PCM186X_SUPPLY_STATUS_DVDD BIT(2) +#define PCM186X_SUPPLY_STATUS_AVDD BIT(1) +#define PCM186X_SUPPLY_STATUS_LDO BIT(0) + +/* PCM186X_MMAP_STAT_CTRL */ +#define PCM186X_MMAP_STAT_DONE BIT(4) +#define PCM186X_MMAP_STAT_BUSY BIT(2) +#define PCM186X_MMAP_STAT_R_REQ BIT(1) +#define PCM186X_MMAP_STAT_W_REQ BIT(0) + +extern const struct regmap_config pcm186x_regmap; + +int pcm186x_probe(struct device *dev, enum pcm186x_type type, int irq, + struct regmap *regmap); +int pcm186x_remove(struct device *dev); + +#endif /* _PCM186X_H_ */ diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index 25c63510ae15..7cdd2dc4fd79 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -70,3 +70,7 @@ static struct spi_driver pcm512x_spi_driver = { }; module_spi_driver(pcm512x_spi_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - SPI"); +MODULE_AUTHOR("Mark Brown <broonie@kernel.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 2df91db765ac..64bf26cec20d 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -289,6 +289,8 @@ static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) dev_err(&rt5514_spi->dev, "%s Failed to reguest IRQ: %d\n", __func__, ret); + else + device_init_wakeup(rt5514_dsp->dev, true); } return 0; @@ -456,8 +458,6 @@ static int rt5514_spi_probe(struct spi_device *spi) return ret; } - device_init_wakeup(&spi->dev, true); - return 0; } @@ -482,10 +482,13 @@ static int __maybe_unused rt5514_resume(struct device *dev) if (device_may_wakeup(dev)) disable_irq_wake(irq); - if (rt5514_dsp->substream) { - rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); - if (buf[0] & RT5514_IRQ_STATUS_BIT) - rt5514_schedule_copy(rt5514_dsp); + if (rt5514_dsp) { + if (rt5514_dsp->substream) { + rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, + sizeof(buf)); + if (buf[0] & RT5514_IRQ_STATUS_BIT) + rt5514_schedule_copy(rt5514_dsp); + } } return 0; diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2a5b5d74e697..2dd6e9f990a4 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -496,7 +496,7 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SUPPLY_S("DMIC CLK", 1, SND_SOC_NOPM, 0, 0, rt5514_set_dmic_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("ADC CLK", RT5514_CLK_CTRL1, diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f020d2d1eef4..edc152c8a1fe 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3823,6 +3823,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, regmap_read(regmap, RT5645_VENDOR_ID, &val); rt5645->v_id = val & 0xff; + regmap_write(rt5645->regmap, RT5645_AD_DA_MIXER, 0x8080); + ret = regmap_register_patch(rt5645->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index b036c9dc0c8c..d329bf719d80 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -1560,6 +1560,10 @@ static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) RT5663_IRQ_POW_SAV_MASK, RT5663_IRQ_POW_SAV_EN); snd_soc_update_bits(codec, RT5663_IRQ_1, RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_RST); + snd_soc_update_bits(codec, RT5663_EM_JACK_TYPE_1, + RT5663_EM_JD_MASK, RT5663_EM_JD_NOR); while (true) { regmap_read(rt5663->regmap, RT5663_INT_ST_2, &val); diff --git a/sound/soc/codecs/rt5663.h b/sound/soc/codecs/rt5663.h index c5a9b69579ad..03adc8004ba9 100644 --- a/sound/soc/codecs/rt5663.h +++ b/sound/soc/codecs/rt5663.h @@ -1029,6 +1029,10 @@ #define RT5663_POL_EXT_JD_SHIFT 10 #define RT5663_POL_EXT_JD_EN (0x1 << 10) #define RT5663_POL_EXT_JD_DIS (0x0 << 10) +#define RT5663_EM_JD_MASK (0x1 << 7) +#define RT5663_EM_JD_SHIFT 7 +#define RT5663_EM_JD_NOR (0x1 << 7) +#define RT5663_EM_JD_RST (0x0 << 7) /* DACREF LDO Control (0x0112)*/ #define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c deleted file mode 100644 index 887923e68849..000000000000 --- a/sound/soc/codecs/sn95031.c +++ /dev/null @@ -1,936 +0,0 @@ -/* - * sn95031.c - TI sn95031 Codec driver - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul <vinod.koul@intel.com> - * Author: Harsha Priya <priya.harsha@intel.com> - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include <linux/platform_device.h> -#include <linux/delay.h> -#include <linux/slab.h> -#include <linux/module.h> - -#include <asm/intel_scu_ipc.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> -#include <sound/jack.h> -#include "sn95031.h" - -#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) -#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) - -/* adc helper functions */ - -/* enables mic bias voltage */ -static void sn95031_enable_mic_bias(struct snd_soc_codec *codec) -{ - snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); - snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); -} - -/* Enable/Disable the ADC depending on the argument */ -static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) -{ - int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); - - if (val) { - /* Enable and start the ADC */ - value |= (SN95031_ADC_ENBL | SN95031_ADC_START); - value &= (~SN95031_ADC_NO_LOOP); - } else { - /* Just stop the ADC */ - value &= (~SN95031_ADC_START); - } - snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); -} - -/* - * finds an empty channel for conversion - * If the ADC is not enabled then start using 0th channel - * itself. Otherwise find an empty channel by looking for a - * channel in which the stopbit is set to 1. returns the index - * of the first free channel if succeeds or an error code. - * - * Context: can sleep - * - */ -static int find_free_channel(struct snd_soc_codec *sn95031_codec) -{ - int i, value; - - /* check whether ADC is enabled */ - value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); - - if ((value & SN95031_ADC_ENBL) == 0) - return 0; - - /* ADC is already enabled; Looking for an empty channel */ - for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { - value = snd_soc_read(sn95031_codec, - SN95031_ADC_CHNL_START_ADDR + i); - if (value & SN95031_STOPBIT_MASK) - break; - } - return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i; -} - -/* Initialize the ADC for reading micbias values. Can sleep. */ -static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) -{ - int base_addr, chnl_addr; - int value; - int channel_index; - - /* Index of the first channel in which the stop bit is set */ - channel_index = find_free_channel(sn95031_codec); - if (channel_index < 0) { - pr_err("No free ADC channels"); - return channel_index; - } - - base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index; - - if (!(channel_index == 0 || channel_index == SN95031_ADC_LOOP_MAX)) { - /* Reset stop bit for channels other than 0 and 12 */ - value = snd_soc_read(sn95031_codec, base_addr); - /* Set the stop bit to zero */ - snd_soc_write(sn95031_codec, base_addr, value & 0xEF); - /* Index of the first free channel */ - base_addr++; - channel_index++; - } - - /* Since this is the last channel, set the stop bit - to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ - snd_soc_write(sn95031_codec, base_addr, - SN95031_AUDIO_DETECT_CODE | 0x10); - - chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index; - pr_debug("mid_initialize : %x", chnl_addr); - configure_adc(sn95031_codec, 1); - return chnl_addr; -} - - -/* reads the ADC registers and gets the mic bias value in mV. */ -static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) -{ - u16 adc_adr = sn95031_initialize_adc(codec); - u16 adc_val1, adc_val2; - unsigned int mic_bias; - - sn95031_enable_mic_bias(codec); - - /* Enable the sound card for conversion before reading */ - snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); - /* Re-toggle the RRDATARD bit */ - snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); - - /* Read the higher bits of data */ - msleep(1000); - adc_val1 = snd_soc_read(codec, adc_adr); - adc_adr++; - adc_val2 = snd_soc_read(codec, adc_adr); - - /* Adding lower two bits to the higher bits */ - mic_bias = (adc_val1 << 2) + (adc_val2 & 3); - mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000; - pr_debug("mic bias = %dmV\n", mic_bias); - return mic_bias; -} -/*end - adc helper functions */ - -static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) -{ - u8 value = 0; - int ret; - - ret = intel_scu_ipc_ioread8(reg, &value); - if (ret == 0) - *val = value; - - return ret; -} - -static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) -{ - return intel_scu_ipc_iowrite8(reg, value); -} - -static const struct regmap_config sn95031_regmap = { - .reg_read = sn95031_read, - .reg_write = sn95031_write, -}; - -static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { - pr_debug("vaud_bias powering up pll\n"); - /* power up the pll */ - snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5)); - /* enable pcm 2 */ - snd_soc_update_bits(codec, SN95031_PCM2C2, - BIT(0), BIT(0)); - } - break; - - case SND_SOC_BIAS_STANDBY: - switch (snd_soc_codec_get_bias_level(codec)) { - case SND_SOC_BIAS_OFF: - pr_debug("vaud_bias power up rail\n"); - /* power up the rail */ - snd_soc_write(codec, SN95031_VAUD, - BIT(2)|BIT(1)|BIT(0)); - msleep(1); - break; - case SND_SOC_BIAS_PREPARE: - /* turn off pcm */ - pr_debug("vaud_bias power dn pcm\n"); - snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0); - snd_soc_write(codec, SN95031_AUDPLLCTRL, 0); - break; - default: - break; - } - break; - - - case SND_SOC_BIAS_OFF: - pr_debug("vaud_bias _OFF doing rail shutdown\n"); - snd_soc_write(codec, SN95031_VAUD, BIT(3)); - break; - } - - return 0; -} - -static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - - if (SND_SOC_DAPM_EVENT_ON(event)) { - pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); - /* power up the rail */ - snd_soc_write(codec, SN95031_VHSP, 0x3D); - snd_soc_write(codec, SN95031_VHSN, 0x3F); - msleep(1); - } else if (SND_SOC_DAPM_EVENT_OFF(event)) { - pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); - snd_soc_write(codec, SN95031_VHSP, 0xC4); - snd_soc_write(codec, SN95031_VHSN, 0x04); - } - return 0; -} - -static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - - if (SND_SOC_DAPM_EVENT_ON(event)) { - pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); - /* power up the rail */ - snd_soc_write(codec, SN95031_VIHF, 0x27); - msleep(1); - } else if (SND_SOC_DAPM_EVENT_OFF(event)) { - pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); - snd_soc_write(codec, SN95031_VIHF, 0x24); - } - return 0; -} - -static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int ldo = 0, clk_dir = 0, data_dir = 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ldo = BIT(5)|BIT(4); - clk_dir = BIT(0); - data_dir = BIT(7); - } - /* program DMIC LDO, clock and set clock */ - snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); - snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(0), clk_dir); - snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(7), data_dir); - return 0; -} - -static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int ldo = 0, clk_dir = 0, data_dir = 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ldo = BIT(5)|BIT(4); - clk_dir = BIT(2); - data_dir = BIT(1); - } - /* program DMIC LDO, clock and set clock */ - snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); - snd_soc_update_bits(codec, SN95031_DMICBUF0123, BIT(2), clk_dir); - snd_soc_update_bits(codec, SN95031_DMICBUF45, BIT(1), data_dir); - return 0; -} - -static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int ldo = 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - ldo = BIT(7)|BIT(6); - - /* program DMIC LDO */ - snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); - return 0; -} - -/* mux controls */ -static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; - -static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum, - SN95031_ADCCONFIG, 1, sn95031_mic_texts); - -static const struct snd_kcontrol_new sn95031_micl_mux_control = - SOC_DAPM_ENUM("Route", sn95031_micl_enum); - -static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum, - SN95031_ADCCONFIG, 3, sn95031_mic_texts); - -static const struct snd_kcontrol_new sn95031_micr_mux_control = - SOC_DAPM_ENUM("Route", sn95031_micr_enum); - -static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", - "DMIC4", "DMIC5", "DMIC6", - "ADC Left", "ADC Right" }; - -static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum, - SN95031_AUDIOMUX12, 0, sn95031_input_texts); - -static const struct snd_kcontrol_new sn95031_input1_mux_control = - SOC_DAPM_ENUM("Route", sn95031_input1_enum); - -static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum, - SN95031_AUDIOMUX12, 4, sn95031_input_texts); - -static const struct snd_kcontrol_new sn95031_input2_mux_control = - SOC_DAPM_ENUM("Route", sn95031_input2_enum); - -static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum, - SN95031_AUDIOMUX34, 0, sn95031_input_texts); - -static const struct snd_kcontrol_new sn95031_input3_mux_control = - SOC_DAPM_ENUM("Route", sn95031_input3_enum); - -static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum, - SN95031_AUDIOMUX34, 4, sn95031_input_texts); - -static const struct snd_kcontrol_new sn95031_input4_mux_control = - SOC_DAPM_ENUM("Route", sn95031_input4_enum); - -/* capture path controls */ - -static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; - -/* 0dB to 30dB in 10dB steps */ -static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0); - -static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum, - SN95031_MICAMP1, 1, sn95031_micmode_text); -static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum, - SN95031_MICAMP2, 1, sn95031_micmode_text); - -static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; - -static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum, - SN95031_DMICMUX, 0, sn95031_dmic_cfg_text); -static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum, - SN95031_DMICMUX, 1, sn95031_dmic_cfg_text); -static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum, - SN95031_DMICMUX, 2, sn95031_dmic_cfg_text); - -static const struct snd_kcontrol_new sn95031_snd_controls[] = { - SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), - SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), - SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), - SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), - SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), - SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, - 2, 4, 0, mic_tlv), - SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, - 2, 4, 0, mic_tlv), -}; - -/* DAPM widgets */ -static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { - - /* all end points mic, hs etc */ - SND_SOC_DAPM_OUTPUT("HPOUTL"), - SND_SOC_DAPM_OUTPUT("HPOUTR"), - SND_SOC_DAPM_OUTPUT("EPOUT"), - SND_SOC_DAPM_OUTPUT("IHFOUTL"), - SND_SOC_DAPM_OUTPUT("IHFOUTR"), - SND_SOC_DAPM_OUTPUT("LINEOUTL"), - SND_SOC_DAPM_OUTPUT("LINEOUTR"), - SND_SOC_DAPM_OUTPUT("VIB1OUT"), - SND_SOC_DAPM_OUTPUT("VIB2OUT"), - - SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */ - SND_SOC_DAPM_INPUT("AMIC2"), - SND_SOC_DAPM_INPUT("DMIC1"), - SND_SOC_DAPM_INPUT("DMIC2"), - SND_SOC_DAPM_INPUT("DMIC3"), - SND_SOC_DAPM_INPUT("DMIC4"), - SND_SOC_DAPM_INPUT("DMIC5"), - SND_SOC_DAPM_INPUT("DMIC6"), - SND_SOC_DAPM_INPUT("LINEINL"), - SND_SOC_DAPM_INPUT("LINEINR"), - - SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0), - SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0), - SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0), - SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0), - SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0), - - SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0, - sn95031_dmic12_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0, - sn95031_dmic34_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0, - sn95031_dmic56_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - - SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, - SND_SOC_NOPM, 0, 0), - - SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, - sn95031_vhs_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, - sn95031_vihf_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - - /* playback path driver enables */ - SND_SOC_DAPM_PGA("Headset Left Playback", - SN95031_DRIVEREN, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("Headset Right Playback", - SN95031_DRIVEREN, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Speaker Left Playback", - SN95031_DRIVEREN, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("Speaker Right Playback", - SN95031_DRIVEREN, 3, 0, NULL, 0), - SND_SOC_DAPM_PGA("Vibra1 Playback", - SN95031_DRIVEREN, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Vibra2 Playback", - SN95031_DRIVEREN, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("Earpiece Playback", - SN95031_DRIVEREN, 6, 0, NULL, 0), - SND_SOC_DAPM_PGA("Lineout Left Playback", - SN95031_LOCTL, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("Lineout Right Playback", - SN95031_LOCTL, 4, 0, NULL, 0), - - /* playback path filter enable */ - SND_SOC_DAPM_PGA("Headset Left Filter", - SN95031_HSEPRXCTRL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Headset Right Filter", - SN95031_HSEPRXCTRL, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("Speaker Left Filter", - SN95031_IHFRXCTRL, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("Speaker Right Filter", - SN95031_IHFRXCTRL, 1, 0, NULL, 0), - - /* DACs */ - SND_SOC_DAPM_DAC("HSDAC Left", "Headset", - SN95031_DACCONFIG, 0, 0), - SND_SOC_DAPM_DAC("HSDAC Right", "Headset", - SN95031_DACCONFIG, 1, 0), - SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker", - SN95031_DACCONFIG, 2, 0), - SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker", - SN95031_DACCONFIG, 3, 0), - SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1", - SN95031_VIB1C5, 1, 0), - SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", - SN95031_VIB2C5, 1, 0), - - /* capture widgets */ - SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1, - 7, 0, NULL, 0), - SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2, - 7, 0, NULL, 0), - - SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0), - SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0), - - /* ADC have null stream as they will be turned ON by TX path */ - SND_SOC_DAPM_ADC("ADC Left", NULL, - SN95031_ADCCONFIG, 0, 0), - SND_SOC_DAPM_ADC("ADC Right", NULL, - SN95031_ADCCONFIG, 2, 0), - - SND_SOC_DAPM_MUX("Mic_InputL Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), - SND_SOC_DAPM_MUX("Mic_InputR Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), - - SND_SOC_DAPM_MUX("Txpath1 Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), - SND_SOC_DAPM_MUX("Txpath2 Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), - SND_SOC_DAPM_MUX("Txpath3 Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), - SND_SOC_DAPM_MUX("Txpath4 Capture Route", - SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), - -}; - -static const struct snd_soc_dapm_route sn95031_audio_map[] = { - /* headset and earpiece map */ - { "HPOUTL", NULL, "Headset Rail"}, - { "HPOUTR", NULL, "Headset Rail"}, - { "HPOUTL", NULL, "Headset Left Playback" }, - { "HPOUTR", NULL, "Headset Right Playback" }, - { "EPOUT", NULL, "Earpiece Playback" }, - { "Headset Left Playback", NULL, "Headset Left Filter"}, - { "Headset Right Playback", NULL, "Headset Right Filter"}, - { "Earpiece Playback", NULL, "Headset Left Filter"}, - { "Headset Left Filter", NULL, "HSDAC Left"}, - { "Headset Right Filter", NULL, "HSDAC Right"}, - - /* speaker map */ - { "IHFOUTL", NULL, "Speaker Rail"}, - { "IHFOUTR", NULL, "Speaker Rail"}, - { "IHFOUTL", NULL, "Speaker Left Playback"}, - { "IHFOUTR", NULL, "Speaker Right Playback"}, - { "Speaker Left Playback", NULL, "Speaker Left Filter"}, - { "Speaker Right Playback", NULL, "Speaker Right Filter"}, - { "Speaker Left Filter", NULL, "IHFDAC Left"}, - { "Speaker Right Filter", NULL, "IHFDAC Right"}, - - /* vibra map */ - { "VIB1OUT", NULL, "Vibra1 Playback"}, - { "Vibra1 Playback", NULL, "Vibra1 DAC"}, - - { "VIB2OUT", NULL, "Vibra2 Playback"}, - { "Vibra2 Playback", NULL, "Vibra2 DAC"}, - - /* lineout */ - { "LINEOUTL", NULL, "Lineout Left Playback"}, - { "LINEOUTR", NULL, "Lineout Right Playback"}, - { "Lineout Left Playback", NULL, "Headset Left Filter"}, - { "Lineout Left Playback", NULL, "Speaker Left Filter"}, - { "Lineout Left Playback", NULL, "Vibra1 DAC"}, - { "Lineout Right Playback", NULL, "Headset Right Filter"}, - { "Lineout Right Playback", NULL, "Speaker Right Filter"}, - { "Lineout Right Playback", NULL, "Vibra2 DAC"}, - - /* Headset (AMIC1) mic */ - { "AMIC1Bias", NULL, "AMIC1"}, - { "MIC1 Enable", NULL, "AMIC1Bias"}, - { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, - - /* AMIC2 */ - { "AMIC2Bias", NULL, "AMIC2"}, - { "MIC2 Enable", NULL, "AMIC2Bias"}, - { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, - - - /* Linein */ - { "LineIn Enable Left", NULL, "LINEINL"}, - { "LineIn Enable Right", NULL, "LINEINR"}, - { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, - { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, - - /* ADC connection */ - { "ADC Left", NULL, "Mic_InputL Capture Route"}, - { "ADC Right", NULL, "Mic_InputR Capture Route"}, - - /*DMIC connections */ - { "DMIC1", NULL, "DMIC12supply"}, - { "DMIC2", NULL, "DMIC12supply"}, - { "DMIC3", NULL, "DMIC34supply"}, - { "DMIC4", NULL, "DMIC34supply"}, - { "DMIC5", NULL, "DMIC56supply"}, - { "DMIC6", NULL, "DMIC56supply"}, - - { "DMIC12Bias", NULL, "DMIC1"}, - { "DMIC12Bias", NULL, "DMIC2"}, - { "DMIC34Bias", NULL, "DMIC3"}, - { "DMIC34Bias", NULL, "DMIC4"}, - { "DMIC56Bias", NULL, "DMIC5"}, - { "DMIC56Bias", NULL, "DMIC6"}, - - /*TX path inputs*/ - { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, - { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, - { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, - { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, - { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath1 Capture Route", "DMIC1", "DMIC1"}, - { "Txpath2 Capture Route", "DMIC1", "DMIC1"}, - { "Txpath3 Capture Route", "DMIC1", "DMIC1"}, - { "Txpath4 Capture Route", "DMIC1", "DMIC1"}, - { "Txpath1 Capture Route", "DMIC2", "DMIC2"}, - { "Txpath2 Capture Route", "DMIC2", "DMIC2"}, - { "Txpath3 Capture Route", "DMIC2", "DMIC2"}, - { "Txpath4 Capture Route", "DMIC2", "DMIC2"}, - { "Txpath1 Capture Route", "DMIC3", "DMIC3"}, - { "Txpath2 Capture Route", "DMIC3", "DMIC3"}, - { "Txpath3 Capture Route", "DMIC3", "DMIC3"}, - { "Txpath4 Capture Route", "DMIC3", "DMIC3"}, - { "Txpath1 Capture Route", "DMIC4", "DMIC4"}, - { "Txpath2 Capture Route", "DMIC4", "DMIC4"}, - { "Txpath3 Capture Route", "DMIC4", "DMIC4"}, - { "Txpath4 Capture Route", "DMIC4", "DMIC4"}, - { "Txpath1 Capture Route", "DMIC5", "DMIC5"}, - { "Txpath2 Capture Route", "DMIC5", "DMIC5"}, - { "Txpath3 Capture Route", "DMIC5", "DMIC5"}, - { "Txpath4 Capture Route", "DMIC5", "DMIC5"}, - { "Txpath1 Capture Route", "DMIC6", "DMIC6"}, - { "Txpath2 Capture Route", "DMIC6", "DMIC6"}, - { "Txpath3 Capture Route", "DMIC6", "DMIC6"}, - { "Txpath4 Capture Route", "DMIC6", "DMIC6"}, - - /* tx path */ - { "TX1 Enable", NULL, "Txpath1 Capture Route"}, - { "TX2 Enable", NULL, "Txpath2 Capture Route"}, - { "TX3 Enable", NULL, "Txpath3 Capture Route"}, - { "TX4 Enable", NULL, "Txpath4 Capture Route"}, - { "PCM_Out", NULL, "TX1 Enable"}, - { "PCM_Out", NULL, "TX2 Enable"}, - { "PCM_Out", NULL, "TX3 Enable"}, - { "PCM_Out", NULL, "TX4 Enable"}, - -}; - -/* speaker and headset mutes, for audio pops and clicks */ -static int sn95031_pcm_hs_mute(struct snd_soc_dai *dai, int mute) -{ - snd_soc_update_bits(dai->codec, - SN95031_HSLVOLCTRL, BIT(7), (!mute << 7)); - snd_soc_update_bits(dai->codec, - SN95031_HSRVOLCTRL, BIT(7), (!mute << 7)); - return 0; -} - -static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) -{ - snd_soc_update_bits(dai->codec, - SN95031_IHFLVOLCTRL, BIT(7), (!mute << 7)); - snd_soc_update_bits(dai->codec, - SN95031_IHFRVOLCTRL, BIT(7), (!mute << 7)); - return 0; -} - -static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) -{ - unsigned int format, rate; - - switch (params_width(params)) { - case 16: - format = BIT(4)|BIT(5); - break; - - case 24: - format = 0; - break; - default: - return -EINVAL; - } - snd_soc_update_bits(dai->codec, SN95031_PCM2C2, - BIT(4)|BIT(5), format); - - switch (params_rate(params)) { - case 48000: - pr_debug("RATE_48000\n"); - rate = 0; - break; - - case 44100: - pr_debug("RATE_44100\n"); - rate = BIT(7); - break; - - default: - pr_err("ERR rate %d\n", params_rate(params)); - return -EINVAL; - } - snd_soc_update_bits(dai->codec, SN95031_PCM1C1, BIT(7), rate); - - return 0; -} - -/* Codec DAI section */ -static const struct snd_soc_dai_ops sn95031_headset_dai_ops = { - .digital_mute = sn95031_pcm_hs_mute, - .hw_params = sn95031_pcm_hw_params, -}; - -static const struct snd_soc_dai_ops sn95031_speaker_dai_ops = { - .digital_mute = sn95031_pcm_spkr_mute, - .hw_params = sn95031_pcm_hw_params, -}; - -static const struct snd_soc_dai_ops sn95031_vib1_dai_ops = { - .hw_params = sn95031_pcm_hw_params, -}; - -static const struct snd_soc_dai_ops sn95031_vib2_dai_ops = { - .hw_params = sn95031_pcm_hw_params, -}; - -static struct snd_soc_dai_driver sn95031_dais[] = { -{ - .name = "SN95031 Headset", - .playback = { - .stream_name = "Headset", - .channels_min = 2, - .channels_max = 2, - .rates = SN95031_RATES, - .formats = SN95031_FORMATS, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 5, - .rates = SN95031_RATES, - .formats = SN95031_FORMATS, - }, - .ops = &sn95031_headset_dai_ops, -}, -{ .name = "SN95031 Speaker", - .playback = { - .stream_name = "Speaker", - .channels_min = 2, - .channels_max = 2, - .rates = SN95031_RATES, - .formats = SN95031_FORMATS, - }, - .ops = &sn95031_speaker_dai_ops, -}, -{ .name = "SN95031 Vibra1", - .playback = { - .stream_name = "Vibra1", - .channels_min = 1, - .channels_max = 1, - .rates = SN95031_RATES, - .formats = SN95031_FORMATS, - }, - .ops = &sn95031_vib1_dai_ops, -}, -{ .name = "SN95031 Vibra2", - .playback = { - .stream_name = "Vibra2", - .channels_min = 1, - .channels_max = 1, - .rates = SN95031_RATES, - .formats = SN95031_FORMATS, - }, - .ops = &sn95031_vib2_dai_ops, -}, -}; - -static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec) -{ - snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); -} - -static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) -{ - snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); - snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); -} - -static int sn95031_get_headset_state(struct snd_soc_codec *codec, - struct snd_soc_jack *mfld_jack) -{ - int micbias = sn95031_get_mic_bias(codec); - - int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); - - pr_debug("jack type detected = %d\n", jack_type); - if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(codec); - return jack_type; -} - -void sn95031_jack_detection(struct snd_soc_codec *codec, - struct mfld_jack_data *jack_data) -{ - unsigned int status; - unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; - - pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id); - if (jack_data->intr_id & 0x1) { - pr_debug("short_push detected\n"); - status = SND_JACK_HEADSET | SND_JACK_BTN_0; - } else if (jack_data->intr_id & 0x2) { - pr_debug("long_push detected\n"); - status = SND_JACK_HEADSET | SND_JACK_BTN_1; - } else if (jack_data->intr_id & 0x4) { - pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(codec, jack_data->mfld_jack); - } else if (jack_data->intr_id & 0x8) { - pr_debug("headset or headphones removed\n"); - status = 0; - sn95031_disable_jack_btn(codec); - } else { - pr_err("unidentified interrupt\n"); - return; - } - - snd_soc_jack_report(jack_data->mfld_jack, status, mask); - /*button pressed and released so we send explicit button release */ - if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1)) - snd_soc_jack_report(jack_data->mfld_jack, - SND_JACK_HEADSET, mask); -} -EXPORT_SYMBOL_GPL(sn95031_jack_detection); - -/* codec registration */ -static int sn95031_codec_probe(struct snd_soc_codec *codec) -{ - pr_debug("codec_probe called\n"); - - /* PCM interface config - * This sets the pcm rx slot conguration to max 6 slots - * for max 4 dais (2 stereo and 2 mono) - */ - snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); - snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); - snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); - snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); - snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); - /* pcm port setting - * This sets the pcm port to slave and clock at 19.2Mhz which - * can support 6slots, sampling rate set per stream in hw-params - */ - snd_soc_write(codec, SN95031_PCM1C1, 0x00); - snd_soc_write(codec, SN95031_PCM2C1, 0x01); - snd_soc_write(codec, SN95031_PCM2C2, 0x0A); - snd_soc_write(codec, SN95031_HSMIXER, BIT(0)|BIT(4)); - /* vendor vibra workround, the vibras are muted by - * custom register so unmute them - */ - snd_soc_write(codec, SN95031_SSR5, 0x80); - snd_soc_write(codec, SN95031_SSR6, 0x80); - snd_soc_write(codec, SN95031_VIB1C5, 0x00); - snd_soc_write(codec, SN95031_VIB2C5, 0x00); - /* configure vibras for pcm port */ - snd_soc_write(codec, SN95031_VIB1C3, 0x00); - snd_soc_write(codec, SN95031_VIB2C3, 0x00); - - /* soft mute ramp time */ - snd_soc_write(codec, SN95031_SOFTMUTE, 0x3); - /* fix the initial volume at 1dB, - * default in +9dB, - * 1dB give optimal swing on DAC, amps - */ - snd_soc_write(codec, SN95031_HSLVOLCTRL, 0x08); - snd_soc_write(codec, SN95031_HSRVOLCTRL, 0x08); - snd_soc_write(codec, SN95031_IHFLVOLCTRL, 0x08); - snd_soc_write(codec, SN95031_IHFRVOLCTRL, 0x08); - /* dac mode and lineout workaround */ - snd_soc_write(codec, SN95031_SSR2, 0x10); - snd_soc_write(codec, SN95031_SSR3, 0x40); - - return 0; -} - -static const struct snd_soc_codec_driver sn95031_codec = { - .probe = sn95031_codec_probe, - .set_bias_level = sn95031_set_vaud_bias, - .idle_bias_off = true, - - .component_driver = { - .controls = sn95031_snd_controls, - .num_controls = ARRAY_SIZE(sn95031_snd_controls), - .dapm_widgets = sn95031_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), - .dapm_routes = sn95031_audio_map, - .num_dapm_routes = ARRAY_SIZE(sn95031_audio_map), - }, -}; - -static int sn95031_device_probe(struct platform_device *pdev) -{ - struct regmap *regmap; - - pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); - - regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); - - return snd_soc_register_codec(&pdev->dev, &sn95031_codec, - sn95031_dais, ARRAY_SIZE(sn95031_dais)); -} - -static int sn95031_device_remove(struct platform_device *pdev) -{ - pr_debug("codec device remove called\n"); - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - -static struct platform_driver sn95031_codec_driver = { - .driver = { - .name = "sn95031", - }, - .probe = sn95031_device_probe, - .remove = sn95031_device_remove, -}; - -module_platform_driver(sn95031_codec_driver); - -MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); -MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>"); -MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sn95031"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h deleted file mode 100644 index 7651fe4e6a45..000000000000 --- a/sound/soc/codecs/sn95031.h +++ /dev/null @@ -1,133 +0,0 @@ -/* - * sn95031.h - TI sn95031 Codec driver - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul <vinod.koul@intel.com> - * Author: Harsha Priya <priya.harsha@intel.com> - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * - */ -#ifndef _SN95031_H -#define _SN95031_H - -/*register map*/ -#define SN95031_VAUD 0xDB -#define SN95031_VHSP 0xDC -#define SN95031_VHSN 0xDD -#define SN95031_VIHF 0xC9 - -#define SN95031_AUDPLLCTRL 0x240 -#define SN95031_DMICBUF0123 0x241 -#define SN95031_DMICBUF45 0x242 -#define SN95031_DMICGPO 0x244 -#define SN95031_DMICMUX 0x245 -#define SN95031_DMICLK 0x246 -#define SN95031_MICBIAS 0x247 -#define SN95031_ADCCONFIG 0x248 -#define SN95031_MICAMP1 0x249 -#define SN95031_MICAMP2 0x24A -#define SN95031_NOISEMUX 0x24B -#define SN95031_AUDIOMUX12 0x24C -#define SN95031_AUDIOMUX34 0x24D -#define SN95031_AUDIOSINC 0x24E -#define SN95031_AUDIOTXEN 0x24F -#define SN95031_HSEPRXCTRL 0x250 -#define SN95031_IHFRXCTRL 0x251 -#define SN95031_HSMIXER 0x256 -#define SN95031_DACCONFIG 0x257 -#define SN95031_SOFTMUTE 0x258 -#define SN95031_HSLVOLCTRL 0x259 -#define SN95031_HSRVOLCTRL 0x25A -#define SN95031_IHFLVOLCTRL 0x25B -#define SN95031_IHFRVOLCTRL 0x25C -#define SN95031_DRIVEREN 0x25D -#define SN95031_LOCTL 0x25E -#define SN95031_VIB1C1 0x25F -#define SN95031_VIB1C2 0x260 -#define SN95031_VIB1C3 0x261 -#define SN95031_VIB1SPIPCM1 0x262 -#define SN95031_VIB1SPIPCM2 0x263 -#define SN95031_VIB1C5 0x264 -#define SN95031_VIB2C1 0x265 -#define SN95031_VIB2C2 0x266 -#define SN95031_VIB2C3 0x267 -#define SN95031_VIB2SPIPCM1 0x268 -#define SN95031_VIB2SPIPCM2 0x269 -#define SN95031_VIB2C5 0x26A -#define SN95031_BTNCTRL1 0x26B -#define SN95031_BTNCTRL2 0x26C -#define SN95031_PCM1TXSLOT01 0x26D -#define SN95031_PCM1TXSLOT23 0x26E -#define SN95031_PCM1TXSLOT45 0x26F -#define SN95031_PCM1RXSLOT0_3 0x270 -#define SN95031_PCM1RXSLOT45 0x271 -#define SN95031_PCM2TXSLOT01 0x272 -#define SN95031_PCM2TXSLOT23 0x273 -#define SN95031_PCM2TXSLOT45 0x274 -#define SN95031_PCM2RXSLOT01 0x275 -#define SN95031_PCM2RXSLOT23 0x276 -#define SN95031_PCM2RXSLOT45 0x277 -#define SN95031_PCM1C1 0x278 -#define SN95031_PCM1C2 0x279 -#define SN95031_PCM1C3 0x27A -#define SN95031_PCM2C1 0x27B -#define SN95031_PCM2C2 0x27C -/*end codec register defn*/ - -/*vendor defn these are not part of avp*/ -#define SN95031_SSR2 0x381 -#define SN95031_SSR3 0x382 -#define SN95031_SSR5 0x384 -#define SN95031_SSR6 0x385 - -/* ADC registers */ - -#define SN95031_ADC1CNTL1 0x1C0 -#define SN95031_ADC_ENBL 0x10 -#define SN95031_ADC_START 0x08 -#define SN95031_ADC1CNTL3 0x1C2 -#define SN95031_ADCTHERM_ENBL 0x04 -#define SN95031_ADCRRDATA_ENBL 0x05 -#define SN95031_STOPBIT_MASK 16 -#define SN95031_ADCTHERM_MASK 4 -#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */ -#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1) -#define SN95031_ADC_NO_LOOP 0x07 -#define SN95031_AUDIO_GPIO_CTRL 0x070 - -/* ADC channel code values */ -#define SN95031_AUDIO_DETECT_CODE 0x06 - -/* ADC base addresses */ -#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ -#define SN95031_ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ -/* multipier to convert to mV */ -#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346 - - -struct mfld_jack_data { - int intr_id; - int micbias_vol; - struct snd_soc_jack *mfld_jack; -}; - -extern void sn95031_jack_detection(struct snd_soc_codec *codec, - struct mfld_jack_data *jack_data); - -#endif diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 730fb2058869..1ff3edb7bbb6 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -116,7 +116,7 @@ struct aic31xx_pdata { /* INT2 interrupt control */ #define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) /* GPIO1 control */ -#define AIC31XX_GPIO1 AIC31XX_REG(0, 50) +#define AIC31XX_GPIO1 AIC31XX_REG(0, 51) #define AIC31XX_DACPRB AIC31XX_REG(0, 60) /* ADC Instruction Set Register */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c482b2e7a7d2..cfe72b9d4356 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -232,7 +232,7 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); struct device_node *twl4030_codec_node = NULL; - twl4030_codec_node = of_find_node_by_name(codec->dev->parent->of_node, + twl4030_codec_node = of_get_child_by_name(codec->dev->parent->of_node, "codec"); if (!pdata && twl4030_codec_node) { @@ -241,9 +241,11 @@ static struct twl4030_codec_data *twl4030_get_pdata(struct snd_soc_codec *codec) GFP_KERNEL); if (!pdata) { dev_err(codec->dev, "Can not allocate memory\n"); + of_node_put(twl4030_codec_node); return NULL; } twl4030_setup_pdata_of(pdata, twl4030_codec_node); + of_node_put(twl4030_codec_node); } return pdata; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 65c059b5ffd7..66e32f5d2917 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1733,7 +1733,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) le64_to_cpu(footer->timestamp)); while (pos < firmware->size && - pos - firmware->size > sizeof(*region)) { + sizeof(*region) < firmware->size - pos) { region = (void *)&(firmware->data[pos]); region_name = "Unknown"; reg = 0; @@ -1782,8 +1782,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); - if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > - firmware->size) { + if (le32_to_cpu(region->len) > + firmware->size - pos - sizeof(*region)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, regions, region_name, @@ -2253,7 +2253,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && - pos - firmware->size > sizeof(*blk)) { + sizeof(*blk) < firmware->size - pos) { blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); @@ -2327,8 +2327,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { - if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > - firmware->size) { + if (le32_to_cpu(blk->len) > + firmware->size - pos - sizeof(*blk)) { adsp_err(dsp, "%s.%d: %s region len %d bytes exceeds file length %zu\n", file, blocks, region_name, diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 0f163abe4ba3..52c27a358933 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -260,8 +260,8 @@ #define ASRFSTi_OUTPUT_FIFO_SHIFT 12 #define ASRFSTi_OUTPUT_FIFO_MASK (((1 << ASRFSTi_OUTPUT_FIFO_WIDTH) - 1) << ASRFSTi_OUTPUT_FIFO_SHIFT) #define ASRFSTi_IAEi_SHIFT 11 -#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_OAFi_SHIFT) -#define ASRFSTi_IAEi (1 << ASRFSTi_OAFi_SHIFT) +#define ASRFSTi_IAEi_MASK (1 << ASRFSTi_IAEi_SHIFT) +#define ASRFSTi_IAEi (1 << ASRFSTi_IAEi_SHIFT) #define ASRFSTi_INPUT_FIFO_WIDTH 7 #define ASRFSTi_INPUT_FIFO_SHIFT 0 #define ASRFSTi_INPUT_FIFO_MASK ((1 << ASRFSTi_INPUT_FIFO_WIDTH) - 1) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f2f51e06e22c..424bafaf51ef 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/ctype.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/slab.h> #include <linux/spinlock.h> #include <linux/of.h> @@ -265,6 +266,8 @@ struct fsl_ssi_private { u32 fifo_watermark; u32 dma_maxburst; + + struct mutex ac97_reg_lock; }; /* @@ -1260,11 +1263,13 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, if (reg > 0x7f) return; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 write clk_prepare_enable failed: %d\n", ret); - return; + goto ret_unlock; } lreg = reg << 12; @@ -1278,6 +1283,9 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); clk_disable_unprepare(fsl_ac97_data->clk); + +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1285,16 +1293,18 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, { struct regmap *regs = fsl_ac97_data->regs; - unsigned short val = -1; + unsigned short val = 0; u32 reg_val; unsigned int lreg; int ret; + mutex_lock(&fsl_ac97_data->ac97_reg_lock); + ret = clk_prepare_enable(fsl_ac97_data->clk); if (ret) { pr_err("ac97 read clk_prepare_enable failed: %d\n", ret); - return -1; + goto ret_unlock; } lreg = (reg & 0x7f) << 12; @@ -1309,6 +1319,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, clk_disable_unprepare(fsl_ac97_data->clk); +ret_unlock: + mutex_unlock(&fsl_ac97_data->ac97_reg_lock); return val; } @@ -1458,12 +1470,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) sizeof(fsl_ssi_ac97_dai)); fsl_ac97_data = ssi_private; - - ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); - if (ret) { - dev_err(&pdev->dev, "could not set AC'97 ops\n"); - return ret; - } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1574,6 +1580,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } + if (fsl_ssi_is_ac97(ssi_private)) { + mutex_init(&ssi_private->ac97_reg_lock); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + goto error_ac97_ops; + } + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, &ssi_private->cpu_dai_drv, 1); if (ret) { @@ -1657,6 +1672,13 @@ error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); error_asoc_register: + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + +error_ac97_ops: + if (fsl_ssi_is_ac97(ssi_private)) + mutex_destroy(&ssi_private->ac97_reg_lock); + if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1675,8 +1697,10 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (fsl_ssi_is_ac97(ssi_private)) + if (fsl_ssi_is_ac97(ssi_private)) { snd_soc_set_ac97_ops(NULL); + mutex_destroy(&ssi_private->ac97_reg_lock); + } return 0; } diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7b49d04e3c60..f2c9e8c5970a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -1,71 +1,122 @@ +config SND_SOC_INTEL_SST_TOPLEVEL + bool "Intel ASoC SST drivers" + default y + depends on X86 || COMPILE_TEST + select SND_SOC_INTEL_MACH + help + Intel ASoC SST Platform Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y + + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel SST drivers. + +if SND_SOC_INTEL_SST_TOPLEVEL + config SND_SST_IPC tristate + # This option controls the IPC core for HiFi2 platforms config SND_SST_IPC_PCI tristate select SND_SST_IPC + # This option controls the PCI-based IPC for HiFi2 platforms + # (Medfield, Merrifield). config SND_SST_IPC_ACPI tristate select SND_SST_IPC - select SND_SOC_INTEL_SST - select IOSF_MBI + # This option controls the ACPI-based IPC for HiFi2 platforms + # (Baytrail, Cherrytrail) -config SND_SOC_INTEL_COMMON +config SND_SOC_INTEL_SST_ACPI tristate + # This option controls ACPI-based probing on + # Haswell/Broadwell/Baytrail legacy and will be set + # when these platforms are enabled config SND_SOC_INTEL_SST tristate - select SND_SOC_INTEL_SST_ACPI if ACPI config SND_SOC_INTEL_SST_FIRMWARE tristate select DW_DMAC_CORE - -config SND_SOC_INTEL_SST_ACPI - tristate - -config SND_SOC_ACPI_INTEL_MATCH - tristate - select SND_SOC_ACPI if ACPI - -config SND_SOC_INTEL_SST_TOPLEVEL - tristate "Intel ASoC SST drivers" - depends on X86 || COMPILE_TEST - select SND_SOC_INTEL_MACH - select SND_SOC_INTEL_COMMON - help - Intel ASoC Audio Drivers. If you have a Intel machine that - has audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y or M - If unsure select "N". + # This option controls firmware download on + # Haswell/Broadwell/Baytrail legacy and will be set + # when these platforms are enabled config SND_SOC_INTEL_HASWELL - tristate "Intel ASoC SST driver for Haswell/Broadwell" - depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF - depends on DMADEVICES + tristate "Haswell/Broadwell Platforms" + depends on SND_DMA_SGBUF + depends on DMADEVICES && ACPI select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_ACPI select SND_SOC_INTEL_SST_FIRMWARE + select SND_SOC_ACPI_INTEL_MATCH + help + If you have a Intel Haswell or Broadwell platform connected to + an I2S codec, then enable this option by saying Y or m. This is + typically used for Chromebooks. This is a recommended option. config SND_SOC_INTEL_BAYTRAIL - tristate "Intel ASoC SST driver for Baytrail (legacy)" - depends on SND_SOC_INTEL_SST_TOPLEVEL - depends on DMADEVICES + tristate "Baytrail (legacy) Platforms" + depends on DMADEVICES && ACPI select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_ACPI select SND_SOC_INTEL_SST_FIRMWARE + select SND_SOC_ACPI_INTEL_MATCH + help + If you have a Intel Baytrail platform connected to an I2S codec, + then enable this option by saying Y or m. This was typically used + for Baytrail Chromebooks but this option is now deprecated and is + not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. + +config SND_SST_ATOM_HIFI2_PLATFORM_PCI + tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + depends on X86 && PCI + select SND_SST_IPC_PCI + select SND_SOC_COMPRESS + help + If you have a Intel Medfield or Merrifield/Edison platform, then + enable this option by saying Y or m. Distros will typically not + enable this option: Medfield devices are not available to + developers and while Merrifield/Edison can run a mainline kernel with + limited functionality it will require a firmware file which + is not in the standard firmware tree config SND_SST_ATOM_HIFI2_PLATFORM - tristate "Intel ASoC SST driver for HiFi2 platforms (*field, *trail)" - depends on SND_SOC_INTEL_SST_TOPLEVEL && X86 + tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + depends on X86 && ACPI + select SND_SST_IPC_ACPI select SND_SOC_COMPRESS + select SND_SOC_ACPI_INTEL_MATCH + select IOSF_MBI + help + If you have a Intel Baytrail or Cherrytrail platform with an I2S + codec, then enable this option by saying Y or m. This is a + recommended option config SND_SOC_INTEL_SKYLAKE - tristate "Intel ASoC SST driver for SKL/BXT/KBL/GLK/CNL" - depends on SND_SOC_INTEL_SST_TOPLEVEL && PCI && ACPI + tristate "SKL/BXT/KBL/GLK/CNL... Platforms" + depends on PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + select SND_SOC_ACPI_INTEL_MATCH + help + If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ + GeminiLake or CannonLake platform with the DSP enabled in the BIOS + then enable this option by saying Y or m. + +config SND_SOC_ACPI_INTEL_MATCH + tristate + select SND_SOC_ACPI if ACPI + # this option controls the compilation of ACPI matching tables and + # helpers and is not meant to be selected by the user. + +endif ## SND_SOC_INTEL_SST_TOPLEVEL # ASoC codec drivers source "sound/soc/intel/boards/Kconfig" diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index b973d457e834..8160520fd74c 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # Core support -obj-$(CONFIG_SND_SOC_INTEL_COMMON) += common/ +obj-$(CONFIG_SND_SOC) += common/ # Platform Support obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += haswell/ diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 32d6e02e2104..6cd481bec275 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -236,6 +236,9 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, ctx->pdata->res_info->acpi_ipc_irq_index); + if (ctx->irq_num <= 0) + return ctx->irq_num < 0 ? ctx->irq_num : -EIO; + return 0; } diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 65e257b17a7e..7ee6aeb7e0af 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -220,10 +220,10 @@ int sst_send_byte_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, sst_free_block(sst_drv_ctx, block); out: test_and_clear_bit(pvt_id, &sst_drv_ctx->pvt_id); - return 0; + return ret; } -/* +/** * sst_pause_stream - Send msg for a pausing stream * @str_id: stream ID * @@ -261,7 +261,7 @@ int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) } } else { retval = -EBADRQC; - dev_dbg(sst_drv_ctx->dev, "SST DBG:BADRQC for stream\n "); + dev_dbg(sst_drv_ctx->dev, "SST DBG:BADRQC for stream\n"); } return retval; @@ -284,7 +284,7 @@ int sst_resume_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) if (!str_info) return -EINVAL; if (str_info->status == STREAM_RUNNING) - return 0; + return 0; if (str_info->status == STREAM_PAUSED) { retval = sst_prepare_and_post_msg(sst_drv_ctx, str_info->task_id, IPC_CMD, IPC_IA_RESUME_STREAM_MRFLD, diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 6f754708a48c..d4e103615f51 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -1,183 +1,183 @@ -config SND_SOC_INTEL_MACH - tristate "Intel Audio machine drivers" +menuconfig SND_SOC_INTEL_MACH + bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL - select SND_SOC_ACPI_INTEL_MATCH if ACPI + help + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y + + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH -config SND_MFLD_MACHINE - tristate "SOC Machine Audio driver for Intel Medfield MID platform" - depends on INTEL_SCU_IPC - select SND_SOC_SN95031 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_PCI - help - This adds support for ASoC machine driver for Intel(R) MID Medfield platform - used as alsa device in audio substem in Intel(R) MID devices - Say Y if you have such a device. - If unsure select "N". +if SND_SOC_INTEL_HASWELL config SND_SOC_INTEL_HASWELL_MACH - tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" + tristate "Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell - Ultrabook platforms. - Say Y if you have such a device. + Ultrabook platforms. This is a recommended option. + Say Y or m if you have such a device. If unsure select "N". config SND_SOC_INTEL_BDW_RT5677_MACH - tristate "ASoC Audio driver for Intel Broadwell with RT5677 codec" - depends on X86_INTEL_LPSS && GPIOLIB && I2C - depends on SND_SOC_INTEL_HASWELL + tristate "Broadwell with RT5677 codec" + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM && GPIOLIB select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with - the RT5677 audio codec. + the RT5677 audio codec. This is a recommended option. + Say Y or m if you have such a device. + If unsure select "N". config SND_SOC_INTEL_BROADWELL_MACH - tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + tristate "Broadwell Wildcatpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell Ultrabook platforms. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_HASWELL + +if SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_BYT_MAX98090_MACH - tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + tristate "Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on SND_SST_IPC_ACPI = n - depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help This adds audio driver for Intel Baytrail platform based boards - with the MAX98090 audio codec. + with the MAX98090 audio codec. This driver is deprecated, use + SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH instead for better + functionality. config SND_SOC_INTEL_BYT_RT5640_MACH - tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" + tristate "Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on SND_SST_IPC_ACPI = n - depends on SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. This driver is deprecated, use SND_SOC_INTEL_BYTCR_RT5640_MACH instead for better functionality. +endif ## SND_SOC_INTEL_BAYTRAIL + +if SND_SST_ATOM_HIFI2_PLATFORM + config SND_SOC_INTEL_BYTCR_RT5640_MACH - tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" - depends on X86 && I2C && ACPI + tristate "Baytrail and Baytrail-CR with RT5640 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI select SND_SOC_RT5640 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help - This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR - platforms with RT5640 audio codec. - Say Y if you have such a device. - If unsure select "N". + This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR + platforms with RT5640 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_BYTCR_RT5651_MACH - tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" - depends on X86 && I2C && ACPI + tristate "Baytrail and Baytrail-CR with RT5651 codec" + depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI select SND_SOC_RT5651 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help - This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR - platforms with RT5651 audio codec. - Say Y if you have such a device. - If unsure select "N". + This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR + platforms with RT5651 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5672_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + tristate "Cherrytrail & Braswell with RT5672 codec" depends on X86_INTEL_LPSS && I2C && ACPI - select SND_SOC_RT5670 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI + select SND_SOC_ACPI + select SND_SOC_RT5670 help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with RT5672 audio codec. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" + tristate "Cherrytrail & Braswell with RT5645/5650 codec" depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI select SND_SOC_RT5645 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with RT5645/5650 audio codec. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH - tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with MAX98090 & TI codec" + tristate "Cherrytrail & Braswell with MAX98090 & TI codec" depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_MAX98090 select SND_SOC_TS3A227E - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell platforms with MAX98090 audio codec it also can support TI jack chip as aux device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_BYT_CHT_DA7213_MACH - tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with DA7212/7213 codec" + tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI select SND_SOC_DA7213 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail & CherryTrail platforms with DA7212/7213 audio codec. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_BYT_CHT_ES8316_MACH - tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail with ES8316 codec" + tristate "Baytrail & Cherrytrail with ES8316 codec" depends on X86_INTEL_LPSS && I2C && ACPI + select SND_SOC_ACPI select SND_SOC_ES8316 - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for Intel(R) Baytrail & Cherrytrail platforms with ES8316 audio codec. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH - tristate "ASoC Audio driver for Intel Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" + tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" depends on X86_INTEL_LPSS && I2C && ACPI - depends on SND_SST_ATOM_HIFI2_PLATFORM - select SND_SST_IPC_ACPI help This adds support for ASoC machine driver for the MinnowBoard Max or Up boards and provides access to I2S signals on the Low-Speed - connector + connector. This is not a recommended option outside of these cases. + It is not intended to be enabled by distros by default. + Say Y or m if you have such a device. + If unsure select "N". +endif ## SND_SST_ATOM_HIFI2_PLATFORM + +if SND_SOC_INTEL_SKYLAKE + config SND_SOC_INTEL_SKL_RT286_MACH - tristate "ASoC Audio driver for SKL with RT286 I2S mode" - depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE + tristate "SKL with RT286 I2S mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_RT286 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for Skylake platforms with RT286 I2S audio codec. - Say Y if you have such a device. + Say Y or m if you have such a device. If unsure select "N". config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH - tristate "ASoC Audio driver for SKL with NAU88L25 and SSM4567 in I2S Mode" - depends on X86_INTEL_LPSS && I2C - depends on SND_SOC_INTEL_SKYLAKE + tristate "SKL with NAU88L25 and SSM4567 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_NAU8825 select SND_SOC_SSM4567 select SND_SOC_DMIC @@ -185,13 +185,12 @@ config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for NAU88L25 + SSM4567. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH - tristate "ASoC Audio driver for SKL with NAU88L25 and MAX98357A in I2S Mode" - depends on X86_INTEL_LPSS && I2C - depends on SND_SOC_INTEL_SKYLAKE + tristate "SKL with NAU88L25 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_NAU8825 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -199,13 +198,12 @@ config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for NAU88L25 + MAX98357A. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH - tristate "ASoC Audio driver for Broxton with DA7219 and MAX98357A in I2S Mode" - depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE + tristate "Broxton with DA7219 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC @@ -214,13 +212,12 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH help This adds support for ASoC machine driver for Broxton-P platforms with DA7219 + MAX98357A I2S audio codec. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_BXT_RT298_MACH - tristate "ASoC Audio driver for Broxton with RT298 I2S mode" - depends on X86 && ACPI && I2C - depends on SND_SOC_INTEL_SKYLAKE + tristate "Broxton with RT298 I2S mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_RT298 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -228,14 +225,12 @@ config SND_SOC_INTEL_BXT_RT298_MACH help This adds support for ASoC machine driver for Broxton platforms with RT286 I2S audio codec. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH - tristate "ASoC Audio driver for KBL with RT5663 and MAX98927 in I2S Mode" - depends on X86_INTEL_LPSS && I2C - select SND_SOC_INTEL_SST - depends on SND_SOC_INTEL_SKYLAKE + tristate "KBL with RT5663 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5663 select SND_SOC_MAX98927 select SND_SOC_DMIC @@ -243,14 +238,13 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + MAX98927. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "ASoC Audio driver for KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on X86_INTEL_LPSS && I2C && SPI - select SND_SOC_INTEL_SST - depends on SND_SOC_INTEL_SKYLAKE + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + depends on SPI select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI @@ -259,7 +253,8 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y if you have such a device. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +endif ## SND_SOC_INTEL_SKYLAKE -endif +endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index c4d82ad41bd7..2179dedb28ad 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -219,7 +219,7 @@ static struct snd_soc_card bytcht_da7213_card = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char codec_name[SND_ACPI_I2C_ID_LEN]; static int bytcht_da7213_probe(struct platform_device *pdev) { @@ -243,7 +243,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 8088396717e3..305e7f4fe55a 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -232,15 +232,39 @@ static struct snd_soc_card byt_cht_es8316_card = { .fully_routed = true, }; +static char codec_name[SND_ACPI_I2C_ID_LEN]; + static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { - int ret = 0; struct byt_cht_es8316_private *priv; + struct snd_soc_acpi_mach *mach; + const char *i2c_name = NULL; + int dai_index = 0; + int i; + int ret = 0; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC); if (!priv) return -ENOMEM; + mach = (&pdev->dev)->platform_data; + /* fix index of codec dai */ + for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { + if (!strcmp(byt_cht_es8316_dais[i].codec_name, + "i2c-ESSX8316:00")) { + dai_index = i; + break; + } + } + + /* fixup codec name based on HID */ + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); + if (i2c_name) { + snprintf(codec_name, sizeof(codec_name), + "%s%s", "i2c-", i2c_name); + byt_cht_es8316_dais[dai_index].codec_name = codec_name; + } + /* register the soc card */ byt_cht_es8316_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f2c0fc415e52..b6a1cfeec830 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -713,7 +713,7 @@ static struct snd_soc_card byt_rt5640_card = { .fully_routed = true, }; -static char byt_rt5640_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ @@ -762,7 +762,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(byt_rt5640_codec_name, sizeof(byt_rt5640_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index d955836c6870..456526a93dd5 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -38,6 +38,8 @@ enum { BYT_RT5651_DMIC_MAP, BYT_RT5651_IN1_MAP, BYT_RT5651_IN2_MAP, + BYT_RT5651_IN1_IN2_MAP, + BYT_RT5651_IN3_MAP, }; #define BYT_RT5651_MAP(quirk) ((quirk) & GENMASK(7, 0)) @@ -62,6 +64,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk IN1_MAP enabled"); if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN2_MAP) dev_info(dev, "quirk IN2_MAP enabled"); + if (BYT_RT5651_MAP(byt_rt5651_quirk) == BYT_RT5651_IN3_MAP) + dev_info(dev, "quirk IN3_MAP enabled"); if (byt_rt5651_quirk & BYT_RT5651_DMIC_EN) dev_info(dev, "quirk DMIC enabled"); if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) @@ -127,6 +131,7 @@ static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Internal Mic", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -138,6 +143,7 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headset Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "Platform Clock"}, {"Speaker", NULL, "Platform Clock"}, + {"Line In", NULL, "Platform Clock"}, {"AIF1 Playback", NULL, "ssp2 Tx"}, {"ssp2 Tx", NULL, "codec_out0"}, @@ -151,6 +157,9 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "LOUTL"}, {"Speaker", NULL, "LOUTR"}, + {"IN2P", NULL, "Line In"}, + {"IN2N", NULL, "Line In"}, + }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic_map[] = { @@ -171,11 +180,25 @@ static const struct snd_soc_dapm_route byt_rt5651_intmic_in2_map[] = { {"IN2P", NULL, "Internal Mic"}, }; +static const struct snd_soc_dapm_route byt_rt5651_intmic_in1_in2_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Internal Mic"}, + {"IN2P", NULL, "Internal Mic"}, + {"IN3P", NULL, "Headset Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5651_intmic_in3_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN3P", NULL, "Headset Mic"}, + {"IN1P", NULL, "Internal Mic"}, +}; + static const struct snd_kcontrol_new byt_rt5651_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Internal Mic"), SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Line In"), }; static struct snd_soc_jack_pin bytcr_jack_pins[] = { @@ -247,8 +270,16 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), }, - .driver_data = (void *)(BYT_RT5651_DMIC_MAP | - BYT_RT5651_DMIC_EN), + .driver_data = (void *)(BYT_RT5651_IN3_MAP), + }, + { + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ADI"), + DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Turbot"), + }, + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN3_MAP), }, { .callback = byt_rt5651_quirk_cb, @@ -256,7 +287,8 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "KIANO"), DMI_MATCH(DMI_PRODUCT_NAME, "KIANO SlimNote 14.2"), }, - .driver_data = (void *)(BYT_RT5651_IN2_MAP), + .driver_data = (void *)(BYT_RT5651_MCLK_EN | + BYT_RT5651_IN1_IN2_MAP), }, {} }; @@ -281,6 +313,14 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime) custom_map = byt_rt5651_intmic_in2_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_in2_map); break; + case BYT_RT5651_IN1_IN2_MAP: + custom_map = byt_rt5651_intmic_in1_in2_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in1_in2_map); + break; + case BYT_RT5651_IN3_MAP: + custom_map = byt_rt5651_intmic_in3_map; + num_routes = ARRAY_SIZE(byt_rt5651_intmic_in3_map); + break; default: custom_map = byt_rt5651_intmic_dmic_map; num_routes = ARRAY_SIZE(byt_rt5651_intmic_dmic_map); @@ -469,7 +509,7 @@ static struct snd_soc_card byt_rt5651_card = { .fully_routed = true, }; -static char byt_rt5651_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN]; static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { @@ -499,7 +539,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(byt_rt5651_codec_name, sizeof(byt_rt5651_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 18d129caa974..976ea6bf9539 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -49,7 +49,7 @@ struct cht_acpi_card { struct cht_mc_private { struct snd_soc_jack jack; struct cht_acpi_card *acpi_card; - char codec_name[16]; + char codec_name[SND_ACPI_I2C_ID_LEN]; struct clk *mclk; }; @@ -499,7 +499,7 @@ static struct cht_acpi_card snd_soc_cards[] = { {"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650}, }; -static char cht_rt5645_codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */ +static char cht_rt5645_codec_name[SND_ACPI_I2C_ID_LEN]; static char cht_rt5645_codec_aif_name[12]; /* = "rt5645-aif[1|2]" */ static char cht_rt5645_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ @@ -566,7 +566,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } /* fixup codec name based on HID */ - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "%s%s", "i2c-", i2c_name); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index f8f21eee9b2d..c14a52d2f714 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -35,7 +35,7 @@ struct cht_mc_private { struct snd_soc_jack headset; - char codec_name[16]; + char codec_name[SND_ACPI_I2C_ID_LEN]; struct clk *mclk; }; @@ -396,7 +396,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* fixup codec name based on HID */ if (mach) { - i2c_name = snd_soc_acpi_find_name_from_hid(mach->id); + i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1); if (i2c_name) { snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", i2c_name); diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 5e1ea0371c90..3c5160779204 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -76,7 +76,7 @@ static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, } /* set correct codec filter for DAI format and clock config */ - snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); + snd_soc_component_update_bits(codec_dai->component, 0x83, 0xffff, 0x8000); return ret; } diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bcf20f3..bf7014ca486f 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, @@ -225,7 +225,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164f2d43..90ea98f01c4c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, @@ -195,7 +195,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) } jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c deleted file mode 100644 index 6f44acfb4aae..000000000000 --- a/sound/soc/intel/boards/mfld_machine.c +++ /dev/null @@ -1,428 +0,0 @@ -/* - * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform - * - * Copyright (C) 2010 Intel Corp - * Author: Vinod Koul <vinod.koul@intel.com> - * Author: Harsha Priya <priya.harsha@intel.com> - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; version 2 of the License. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt - -#include <linux/init.h> -#include <linux/device.h> -#include <linux/slab.h> -#include <linux/io.h> -#include <linux/module.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/jack.h> -#include "../codecs/sn95031.h" - -#define MID_MONO 1 -#define MID_STEREO 2 -#define MID_MAX_CAP 5 -#define MFLD_JACK_INSERT 0x04 - -enum soc_mic_bias_zones { - MFLD_MV_START = 0, - /* mic bias volutage range for Headphones*/ - MFLD_MV_HP = 400, - /* mic bias volutage range for American Headset*/ - MFLD_MV_AM_HS = 650, - /* mic bias volutage range for Headset*/ - MFLD_MV_HS = 2000, - MFLD_MV_UNDEFINED, -}; - -static unsigned int hs_switch; -static unsigned int lo_dac; -static struct snd_soc_codec *mfld_codec; - -struct mfld_mc_private { - void __iomem *int_base; - u8 interrupt_status; -}; - -struct snd_soc_jack mfld_jack; - -/*Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin mfld_jack_pins[] = { - { - .pin = "Headphones", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "AMIC1", - .mask = SND_JACK_MICROPHONE, - }, -}; - -/* jack detection voltage zones */ -static struct snd_soc_jack_zone mfld_zones[] = { - {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, - {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, -}; - -/* sound card controls */ -static const char * const headset_switch_text[] = {"Earpiece", "Headset"}; - -static const char * const lo_text[] = {"Vibra", "Headset", "IHF", "None"}; - -static const struct soc_enum headset_enum = - SOC_ENUM_SINGLE_EXT(2, headset_switch_text); - -static const struct soc_enum lo_enum = - SOC_ENUM_SINGLE_EXT(4, lo_text); - -static int headset_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = hs_switch; - return 0; -} - -static int headset_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &card->dapm; - - if (ucontrol->value.enumerated.item[0] == hs_switch) - return 0; - - snd_soc_dapm_mutex_lock(dapm); - - if (ucontrol->value.enumerated.item[0]) { - pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - } else { - pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); - } - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); - - hs_switch = ucontrol->value.enumerated.item[0]; - - return 0; -} - -static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) -{ - snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL"); - snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR"); - snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL"); - snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR"); - snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT"); - snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT"); - if (hs_switch) { - snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - } else { - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); - } -} - -static int lo_get_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = lo_dac; - return 0; -} - -static int lo_set_switch(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_context *dapm = &card->dapm; - - if (ucontrol->value.enumerated.item[0] == lo_dac) - return 0; - - snd_soc_dapm_mutex_lock(dapm); - - /* we dont want to work with last state of lineout so just enable all - * pins and then disable pins not required - */ - lo_enable_out_pins(dapm); - - switch (ucontrol->value.enumerated.item[0]) { - case 0: - pr_debug("set vibra path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); - snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0); - break; - - case 1: - pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); - snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22); - break; - - case 2: - pr_debug("set spkr path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL"); - snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44); - break; - - case 3: - pr_debug("set null path\n"); - snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR"); - snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66); - break; - } - - snd_soc_dapm_sync_unlocked(dapm); - - snd_soc_dapm_mutex_unlock(dapm); - - lo_dac = ucontrol->value.enumerated.item[0]; - return 0; -} - -static const struct snd_kcontrol_new mfld_snd_controls[] = { - SOC_ENUM_EXT("Playback Switch", headset_enum, - headset_get_switch, headset_set_switch), - SOC_ENUM_EXT("Lineout Mux", lo_enum, - lo_get_switch, lo_set_switch), -}; - -static const struct snd_soc_dapm_widget mfld_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Mic", NULL), -}; - -static const struct snd_soc_dapm_route mfld_map[] = { - {"Headphones", NULL, "HPOUTR"}, - {"Headphones", NULL, "HPOUTL"}, - {"Mic", NULL, "AMIC1"}, -}; - -static void mfld_jack_check(unsigned int intr_status) -{ - struct mfld_jack_data jack_data; - - if (!mfld_codec) - return; - - jack_data.mfld_jack = &mfld_jack; - jack_data.intr_id = intr_status; - - sn95031_jack_detection(mfld_codec, &jack_data); - /* TODO: add american headset detection post gpiolib support */ -} - -static int mfld_init(struct snd_soc_pcm_runtime *runtime) -{ - struct snd_soc_dapm_context *dapm = &runtime->card->dapm; - int ret_val; - - /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "Headphones"); - /* default is lineout NC, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); - lo_dac = 3; - hs_switch = 0; - /* we dont use linein in this so set to NC */ - snd_soc_dapm_disable_pin(dapm, "LINEINL"); - snd_soc_dapm_disable_pin(dapm, "LINEINR"); - - /* Headset and button jack detection */ - ret_val = snd_soc_card_jack_new(runtime->card, - "Intel(R) MID Audio Jack", SND_JACK_HEADSET | - SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, - mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); - if (ret_val) { - pr_err("jack creation failed\n"); - return ret_val; - } - - ret_val = snd_soc_jack_add_zones(&mfld_jack, - ARRAY_SIZE(mfld_zones), mfld_zones); - if (ret_val) { - pr_err("adding jack zones failed\n"); - return ret_val; - } - - mfld_codec = runtime->codec; - - /* we want to check if anything is inserted at boot, - * so send a fake event to codec and it will read adc - * to find if anything is there or not */ - mfld_jack_check(MFLD_JACK_INSERT); - return ret_val; -} - -static struct snd_soc_dai_link mfld_msic_dailink[] = { - { - .name = "Medfield Headset", - .stream_name = "Headset", - .cpu_dai_name = "Headset-cpu-dai", - .codec_dai_name = "SN95031 Headset", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = mfld_init, - }, - { - .name = "Medfield Speaker", - .stream_name = "Speaker", - .cpu_dai_name = "Speaker-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Vibra", - .stream_name = "Vibra1", - .cpu_dai_name = "Vibra1-cpu-dai", - .codec_dai_name = "SN95031 Vibra1", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Haptics", - .stream_name = "Vibra2", - .cpu_dai_name = "Vibra2-cpu-dai", - .codec_dai_name = "SN95031 Vibra2", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, - { - .name = "Medfield Compress", - .stream_name = "Speaker", - .cpu_dai_name = "Compress-cpu-dai", - .codec_dai_name = "SN95031 Speaker", - .codec_name = "sn95031", - .platform_name = "sst-platform", - .init = NULL, - }, -}; - -/* SoC card */ -static struct snd_soc_card snd_soc_card_mfld = { - .name = "medfield_audio", - .owner = THIS_MODULE, - .dai_link = mfld_msic_dailink, - .num_links = ARRAY_SIZE(mfld_msic_dailink), - - .controls = mfld_snd_controls, - .num_controls = ARRAY_SIZE(mfld_snd_controls), - .dapm_widgets = mfld_widgets, - .num_dapm_widgets = ARRAY_SIZE(mfld_widgets), - .dapm_routes = mfld_map, - .num_dapm_routes = ARRAY_SIZE(mfld_map), -}; - -static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) -{ - struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; - - memcpy_fromio(&mc_private->interrupt_status, - ((void *)(mc_private->int_base)), - sizeof(u8)); - return IRQ_WAKE_THREAD; -} - -static irqreturn_t snd_mfld_jack_detection(int irq, void *data) -{ - struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - - mfld_jack_check(mc_drv_ctx->interrupt_status); - - return IRQ_HANDLED; -} - -static int snd_mfld_mc_probe(struct platform_device *pdev) -{ - int ret_val = 0, irq; - struct mfld_mc_private *mc_drv_ctx; - struct resource *irq_mem; - - pr_debug("snd_mfld_mc_probe called\n"); - - /* retrive the irq number */ - irq = platform_get_irq(pdev, 0); - - /* audio interrupt base of SRAM location where - * interrupts are stored by System FW */ - mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC); - if (!mc_drv_ctx) - return -ENOMEM; - - irq_mem = platform_get_resource_byname( - pdev, IORESOURCE_MEM, "IRQ_BASE"); - if (!irq_mem) { - pr_err("no mem resource given\n"); - return -ENODEV; - } - mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start, - resource_size(irq_mem)); - if (!mc_drv_ctx->int_base) { - pr_err("Mapping of cache failed\n"); - return -ENOMEM; - } - /* register for interrupt */ - ret_val = devm_request_threaded_irq(&pdev->dev, irq, - snd_mfld_jack_intr_handler, - snd_mfld_jack_detection, - IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); - if (ret_val) { - pr_err("cannot register IRQ\n"); - return ret_val; - } - /* register the soc card */ - snd_soc_card_mfld.dev = &pdev->dev; - ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); - if (ret_val) { - pr_debug("snd_soc_register_card failed %d\n", ret_val); - return ret_val; - } - platform_set_drvdata(pdev, mc_drv_ctx); - pr_debug("successfully exited probe\n"); - return 0; -} - -static struct platform_driver snd_mfld_mc_driver = { - .driver = { - .name = "msic_audio", - }, - .probe = snd_mfld_mc_probe, -}; - -module_platform_driver(snd_mfld_mc_driver); - -MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); -MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>"); -MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:msic-audio"); diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index 11c0805393ff..fd82f4b1d4a0 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -269,7 +269,7 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask, */ timeout = jiffies + msecs_to_jiffies(time); - while (((sst_dsp_shim_read_unlocked(ctx, offset) & mask) != target) + while ((((reg = sst_dsp_shim_read_unlocked(ctx, offset)) & mask) != target) && time_before(jiffies, timeout)) { k++; if (k > 10) @@ -278,8 +278,6 @@ int sst_dsp_register_poll(struct sst_dsp *ctx, u32 offset, u32 mask, usleep_range(s, 2*s); } - reg = sst_dsp_shim_read_unlocked(ctx, offset); - if ((reg & mask) == target) { dev_dbg(ctx->dev, "FW Poll Status: reg=%#x %s successful\n", reg, operation); diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 4524211960e4..440bca7afbf1 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -595,7 +595,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, INIT_DELAYED_WORK(&skl->d0i3.work, bxt_set_dsp_D0i3); skl->d0i3.state = SKL_DSP_D0I3_NONE; - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 387de388ce29..245df1067ba8 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -458,7 +458,7 @@ int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, cnl->boot_complete = false; init_waitqueue_head(&cnl->boot_wait); - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(cnl_sst_dsp_init); diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h new file mode 100644 index 000000000000..dcf819bc688f --- /dev/null +++ b/sound/soc/intel/skylake/skl-i2s.h @@ -0,0 +1,64 @@ +/* + * skl-i2s.h - i2s blob mapping + * + * Copyright (C) 2017 Intel Corp + * Author: Subhransu S. Prusty < subhransu.s.prusty@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SOUND_SOC_SKL_I2S_H +#define __SOUND_SOC_SKL_I2S_H + +#define SKL_I2S_MAX_TIME_SLOTS 8 +#define SKL_MCLK_DIV_CLK_SRC_MASK GENMASK(17, 16) + +#define SKL_MNDSS_DIV_CLK_SRC_MASK GENMASK(21, 20) +#define SKL_SHIFT(x) (ffs(x) - 1) +#define SKL_MCLK_DIV_RATIO_MASK GENMASK(11, 0) + +struct skl_i2s_config { + u32 ssc0; + u32 ssc1; + u32 sscto; + u32 sspsp; + u32 sstsa; + u32 ssrsa; + u32 ssc2; + u32 sspsp2; + u32 ssc3; + u32 ssioc; +} __packed; + +struct skl_i2s_config_mclk { + u32 mdivctrl; + u32 mdivr; +}; + +/** + * struct skl_i2s_config_blob_legacy - Structure defines I2S Gateway + * configuration legacy blob + * + * @gtw_attr: Gateway attribute for the I2S Gateway + * @tdm_ts_group: TDM slot mapping against channels in the Gateway. + * @i2s_cfg: I2S HW registers + * @mclk: MCLK clock source and divider values + */ +struct skl_i2s_config_blob_legacy { + u32 gtw_attr; + u32 tdm_ts_group[SKL_I2S_MAX_TIME_SLOTS]; + struct skl_i2s_config i2s_cfg; + struct skl_i2s_config_mclk mclk; +}; + +#endif /* __SOUND_SOC_SKL_I2S_H */ diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 61b5bfa79d13..8cbf080c38b3 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -55,6 +55,19 @@ static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) return 0; } +#define SKL_ASTATE_PARAM_ID 4 + +void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data) +{ + struct skl_ipc_large_config_msg msg = {0}; + + msg.large_param_id = SKL_ASTATE_PARAM_ID; + msg.param_data_size = (cnt * sizeof(struct skl_astate_param) + + sizeof(cnt)); + + skl_ipc_set_large_config(&ctx->ipc, &msg, data); +} + #define NOTIFICATION_PARAM_ID 3 #define NOTIFICATION_MASK 0xf @@ -404,11 +417,20 @@ int skl_resume_dsp(struct skl *skl) if (skl->skl_sst->is_first_boot == true) return 0; + /* disable dynamic clock gating during fw and lib download */ + ctx->enable_miscbdcge(ctx->dev, false); + ret = skl_dsp_wake(ctx->dsp); + ctx->enable_miscbdcge(ctx->dev, true); if (ret < 0) return ret; skl_dsp_enable_notification(skl->skl_sst, false); + + if (skl->cfg.astate_cfg != NULL) { + skl_dsp_set_astate_cfg(skl->skl_sst, skl->cfg.astate_cfg->count, + skl->cfg.astate_cfg); + } return ret; } diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index d14c50a60289..3b1d2b828c1b 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -19,6 +19,7 @@ */ #include <linux/pci.h> #include "skl.h" +#include "skl-i2s.h" #define NHLT_ACPI_HEADER_SIG "NHLT" @@ -43,7 +44,8 @@ struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) obj = acpi_evaluate_dsm(handle, &osc_guid, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - nhlt_table = (struct nhlt_acpi_table *) + if (nhlt_ptr->length) + nhlt_table = (struct nhlt_acpi_table *) memremap(nhlt_ptr->min_addr, nhlt_ptr->length, MEMREMAP_WB); ACPI_FREE(obj); @@ -119,11 +121,16 @@ static bool skl_check_ep_match(struct device *dev, struct nhlt_endpoint *epnt, if ((epnt->virtual_bus_id == instance_id) && (epnt->linktype == link_type) && - (epnt->direction == dirn) && - (epnt->device_type == dev_type)) - return true; - else - return false; + (epnt->direction == dirn)) { + /* do not check dev_type for DMIC link type */ + if (epnt->linktype == NHLT_LINK_DMIC) + return true; + + if (epnt->device_type == dev_type) + return true; + } + + return false; } struct nhlt_specific_cfg @@ -271,3 +278,157 @@ void skl_nhlt_remove_sysfs(struct skl *skl) sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); } + +/* + * Queries NHLT for all the fmt configuration for a particular endpoint and + * stores all possible rates supported in a rate table for the corresponding + * sclk/sclkfs. + */ +static void skl_get_ssp_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks, + struct nhlt_fmt *fmt, u8 id) +{ + struct skl_i2s_config_blob_legacy *i2s_config; + struct skl_clk_parent_src *parent; + struct skl_ssp_clk *sclk, *sclkfs; + struct nhlt_fmt_cfg *fmt_cfg; + struct wav_fmt_ext *wav_fmt; + unsigned long rate = 0; + bool present = false; + int rate_index = 0; + u16 channels, bps; + u8 clk_src; + int i, j; + u32 fs; + + sclk = &ssp_clks[SKL_SCLK_OFS]; + sclkfs = &ssp_clks[SKL_SCLKFS_OFS]; + + if (fmt->fmt_count == 0) + return; + + for (i = 0; i < fmt->fmt_count; i++) { + fmt_cfg = &fmt->fmt_config[i]; + wav_fmt = &fmt_cfg->fmt_ext; + + channels = wav_fmt->fmt.channels; + bps = wav_fmt->fmt.bits_per_sample; + fs = wav_fmt->fmt.samples_per_sec; + + /* + * In case of TDM configuration on a ssp, there can + * be more than one blob in which channel masks are + * different for each usecase for a specific rate and bps. + * But the sclk rate will be generated for the total + * number of channels used for that endpoint. + * + * So for the given fs and bps, choose blob which has + * the superset of all channels for that endpoint and + * derive the rate. + */ + for (j = i; j < fmt->fmt_count; j++) { + fmt_cfg = &fmt->fmt_config[j]; + wav_fmt = &fmt_cfg->fmt_ext; + if ((fs == wav_fmt->fmt.samples_per_sec) && + (bps == wav_fmt->fmt.bits_per_sample)) + channels = max_t(u16, channels, + wav_fmt->fmt.channels); + } + + rate = channels * bps * fs; + + /* check if the rate is added already to the given SSP's sclk */ + for (j = 0; (j < SKL_MAX_CLK_RATES) && + (sclk[id].rate_cfg[j].rate != 0); j++) { + if (sclk[id].rate_cfg[j].rate == rate) { + present = true; + break; + } + } + + /* Fill rate and parent for sclk/sclkfs */ + if (!present) { + /* MCLK Divider Source Select */ + i2s_config = (struct skl_i2s_config_blob_legacy *) + fmt->fmt_config[0].config.caps; + clk_src = ((i2s_config->mclk.mdivctrl) + & SKL_MNDSS_DIV_CLK_SRC_MASK) >> + SKL_SHIFT(SKL_MNDSS_DIV_CLK_SRC_MASK); + + parent = skl_get_parent_clk(clk_src); + + /* + * Do not copy the config data if there is no parent + * clock available for this clock source select + */ + if (!parent) + continue; + + sclk[id].rate_cfg[rate_index].rate = rate; + sclk[id].rate_cfg[rate_index].config = fmt_cfg; + sclkfs[id].rate_cfg[rate_index].rate = rate; + sclkfs[id].rate_cfg[rate_index].config = fmt_cfg; + sclk[id].parent_name = parent->name; + sclkfs[id].parent_name = parent->name; + + rate_index++; + } + } +} + +static void skl_get_mclk(struct skl *skl, struct skl_ssp_clk *mclk, + struct nhlt_fmt *fmt, u8 id) +{ + struct skl_i2s_config_blob_legacy *i2s_config; + struct nhlt_specific_cfg *fmt_cfg; + struct skl_clk_parent_src *parent; + u32 clkdiv, div_ratio; + u8 clk_src; + + fmt_cfg = &fmt->fmt_config[0].config; + i2s_config = (struct skl_i2s_config_blob_legacy *)fmt_cfg->caps; + + /* MCLK Divider Source Select */ + clk_src = ((i2s_config->mclk.mdivctrl) & SKL_MCLK_DIV_CLK_SRC_MASK) >> + SKL_SHIFT(SKL_MCLK_DIV_CLK_SRC_MASK); + + clkdiv = i2s_config->mclk.mdivr & SKL_MCLK_DIV_RATIO_MASK; + + /* bypass divider */ + div_ratio = 1; + + if (clkdiv != SKL_MCLK_DIV_RATIO_MASK) + /* Divider is 2 + clkdiv */ + div_ratio = clkdiv + 2; + + /* Calculate MCLK rate from source using div value */ + parent = skl_get_parent_clk(clk_src); + if (!parent) + return; + + mclk[id].rate_cfg[0].rate = parent->rate/div_ratio; + mclk[id].rate_cfg[0].config = &fmt->fmt_config[0]; + mclk[id].parent_name = parent->name; +} + +void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks) +{ + struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_endpoint *epnt; + struct nhlt_fmt *fmt; + int i; + u8 id; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + for (i = 0; i < nhlt->endpoint_count; i++) { + if (epnt->linktype == NHLT_LINK_SSP) { + id = epnt->virtual_bus_id; + + fmt = (struct nhlt_fmt *)(epnt->config.caps + + epnt->config.size); + + skl_get_ssp_clks(skl, ssp_clks, fmt, id); + skl_get_mclk(skl, ssp_clks, fmt, id); + } + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } +} diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 1dd97479e0c0..e46828533826 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -537,7 +537,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + link = snd_hdac_ext_bus_get_link(ebus, codec_dai->component->name); if (!link) return -EINVAL; @@ -620,7 +620,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - link = snd_hdac_ext_bus_get_link(ebus, rtd->codec->component.name); + link = snd_hdac_ext_bus_get_link(ebus, rtd->codec_dai->component->name); if (!link) return -EINVAL; @@ -1343,7 +1343,11 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform) return -EIO; } + /* disable dynamic clock gating during fw and lib download */ + skl->skl_sst->enable_miscbdcge(platform->dev, false); + ret = ops->init_fw(platform->dev, skl->skl_sst); + skl->skl_sst->enable_miscbdcge(platform->dev, true); if (ret < 0) { dev_err(platform->dev, "Failed to boot first fw: %d\n", ret); return ret; @@ -1351,6 +1355,12 @@ static int skl_platform_soc_probe(struct snd_soc_platform *platform) skl_populate_modules(skl); skl->skl_sst->update_d0i3c = skl_update_d0i3c; skl_dsp_enable_notification(skl->skl_sst, false); + + if (skl->cfg.astate_cfg != NULL) { + skl_dsp_set_astate_cfg(skl->skl_sst, + skl->cfg.astate_cfg->count, + skl->cfg.astate_cfg); + } } pm_runtime_mark_last_busy(platform->dev); pm_runtime_put_autosuspend(platform->dev); diff --git a/sound/soc/intel/skylake/skl-ssp-clk.h b/sound/soc/intel/skylake/skl-ssp-clk.h new file mode 100644 index 000000000000..c9ea84004260 --- /dev/null +++ b/sound/soc/intel/skylake/skl-ssp-clk.h @@ -0,0 +1,79 @@ +/* + * skl-ssp-clk.h - Skylake ssp clock information and ipc structure + * + * Copyright (C) 2017 Intel Corp + * Author: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com> + * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef SOUND_SOC_SKL_SSP_CLK_H +#define SOUND_SOC_SKL_SSP_CLK_H + +#define SKL_MAX_SSP 6 +/* xtal/cardinal/pll, parent of ssp clocks and mclk */ +#define SKL_MAX_CLK_SRC 3 +#define SKL_MAX_SSP_CLK_TYPES 3 /* mclk, sclk, sclkfs */ + +#define SKL_MAX_CLK_CNT (SKL_MAX_SSP * SKL_MAX_SSP_CLK_TYPES) + +/* Max number of configurations supported for each clock */ +#define SKL_MAX_CLK_RATES 10 + +#define SKL_SCLK_OFS SKL_MAX_SSP +#define SKL_SCLKFS_OFS (SKL_SCLK_OFS + SKL_MAX_SSP) + +enum skl_clk_type { + SKL_MCLK, + SKL_SCLK, + SKL_SCLK_FS, +}; + +enum skl_clk_src_type { + SKL_XTAL, + SKL_CARDINAL, + SKL_PLL, +}; + +struct skl_clk_parent_src { + u8 clk_id; + const char *name; + unsigned long rate; + const char *parent_name; +}; + +struct skl_clk_rate_cfg_table { + unsigned long rate; + void *config; +}; + +/* + * rate for mclk will be in rates[0]. For sclk and sclkfs, rates[] store + * all possible clocks ssp can generate for that platform. + */ +struct skl_ssp_clk { + const char *name; + const char *parent_name; + struct skl_clk_rate_cfg_table rate_cfg[SKL_MAX_CLK_RATES]; +}; + +struct skl_clk_pdata { + struct skl_clk_parent_src *parent_clks; + int num_clks; + struct skl_ssp_clk *ssp_clks; + void *pvt_data; +}; + +#endif /* SOUND_SOC_SKL_SSP_CLK_H */ diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index 19ee1d4f3bdf..71e31ad0bb3f 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -435,16 +435,22 @@ struct sst_dsp *skl_dsp_ctx_init(struct device *dev, return NULL; } + return sst; +} + +int skl_dsp_acquire_irq(struct sst_dsp *sst) +{ + struct sst_dsp_device *sst_dev = sst->sst_dev; + int ret; + /* Register the ISR */ ret = request_threaded_irq(sst->irq, sst->ops->irq_handler, sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); - if (ret) { + if (ret) dev_err(sst->dev, "unable to grab threaded IRQ %d, disabling device\n", sst->irq); - return NULL; - } - return sst; + return ret; } void skl_dsp_free(struct sst_dsp *dsp) diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index eba20d37ba8c..12fc9a73dc8a 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -206,6 +206,7 @@ int skl_cldma_wait_interruptible(struct sst_dsp *ctx); void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state); struct sst_dsp *skl_dsp_ctx_init(struct device *dev, struct sst_dsp_device *sst_dev, int irq); +int skl_dsp_acquire_irq(struct sst_dsp *sst); bool is_skl_dsp_running(struct sst_dsp *ctx); unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx); @@ -251,6 +252,9 @@ void skl_freeup_uuid_list(struct skl_sst *ctx); int skl_dsp_strip_extended_manifest(struct firmware *fw); void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable); + +void skl_dsp_set_astate_cfg(struct skl_sst *ctx, u32 cnt, void *data); + int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp, struct sst_dsp_device *skl_dev); diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 8ff89280d9fd..2ae405617876 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -178,7 +178,8 @@ static inline int skl_pvtid_128(struct uuid_module *module) * skl_get_pvt_id: generate a private id for use as module id * * @ctx: driver context - * @mconfig: module configuration data + * @uuid_mod: module's uuid + * @instance_id: module's instance id * * This generates a 128 bit private unique id for a module TYPE so that * module instance is unique @@ -208,7 +209,8 @@ EXPORT_SYMBOL_GPL(skl_get_pvt_id); * skl_put_pvt_id: free up the private id allocated * * @ctx: driver context - * @mconfig: module configuration data + * @uuid_mod: module's uuid + * @pvt_id: module pvt id * * This frees a 128 bit private unique id previously generated */ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index a436abf2fe3f..5a7e41b65ef3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -569,7 +569,7 @@ int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, sst->fw_ops = skl_fw_ops; - return 0; + return skl_dsp_acquire_irq(sst); } EXPORT_SYMBOL_GPL(skl_sst_dsp_init); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a072bcf209d2..73af6e19ebbd 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -190,7 +190,6 @@ skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig) u8 res_idx = mconfig->res_idx; struct skl_module_res *res = &mconfig->module->resources[res_idx]; - res = &mconfig->module->resources[res_idx]; skl->resource.mcps -= res->cps; } @@ -2908,7 +2907,7 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, break; default: - dev_warn(bus->dev, "Control load not supported %d:%d:%d\n", + dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); break; } @@ -3056,11 +3055,13 @@ static int skl_tplg_get_int_tkn(struct device *dev, struct snd_soc_tplg_vendor_value_elem *tkn_elem, struct skl *skl) { - int tkn_count = 0, ret; + int tkn_count = 0, ret, size; static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx; struct skl_module_res *res = NULL; struct skl_module_iface *fmt = NULL; struct skl_module *mod = NULL; + static struct skl_astate_param *astate_table; + static int astate_cfg_idx, count; int i; if (skl->modules) { @@ -3093,6 +3094,46 @@ static int skl_tplg_get_int_tkn(struct device *dev, mod_idx = tkn_elem->value; break; + case SKL_TKN_U32_ASTATE_COUNT: + if (astate_table != NULL) { + dev_err(dev, "More than one entry for A-State count"); + return -EINVAL; + } + + if (tkn_elem->value > SKL_MAX_ASTATE_CFG) { + dev_err(dev, "Invalid A-State count %d\n", + tkn_elem->value); + return -EINVAL; + } + + size = tkn_elem->value * sizeof(struct skl_astate_param) + + sizeof(count); + skl->cfg.astate_cfg = devm_kzalloc(dev, size, GFP_KERNEL); + if (!skl->cfg.astate_cfg) + return -ENOMEM; + + astate_table = skl->cfg.astate_cfg->astate_table; + count = skl->cfg.astate_cfg->count = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_IDX: + if (tkn_elem->value >= count) { + dev_err(dev, "Invalid A-State index %d\n", + tkn_elem->value); + return -EINVAL; + } + + astate_cfg_idx = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_KCPS: + astate_table[astate_cfg_idx].kcps = tkn_elem->value; + break; + + case SKL_TKN_U32_ASTATE_CLK_SRC: + astate_table[astate_cfg_idx].clk_src = tkn_elem->value; + break; + case SKL_TKN_U8_IN_PIN_TYPE: case SKL_TKN_U8_OUT_PIN_TYPE: case SKL_TKN_U8_IN_QUEUE_COUNT: diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..32ce64c6b2dc 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -355,6 +355,7 @@ static int skl_resume(struct device *dev) if (ebus->cmd_dma_state) snd_hdac_bus_init_cmd_io(&ebus->bus); + ret = 0; } else { ret = _skl_resume(ebus); @@ -435,19 +436,51 @@ static int skl_free(struct hdac_ext_bus *ebus) return 0; } -static int skl_machine_device_register(struct skl *skl, void *driver_data) +/* + * For each ssp there are 3 clocks (mclk/sclk/sclkfs). + * e.g. for ssp0, clocks will be named as + * "ssp0_mclk", "ssp0_sclk", "ssp0_sclkfs" + * So for skl+, there are 6 ssps, so 18 clocks will be created. + */ +static struct skl_ssp_clk skl_ssp_clks[] = { + {.name = "ssp0_mclk"}, {.name = "ssp1_mclk"}, {.name = "ssp2_mclk"}, + {.name = "ssp3_mclk"}, {.name = "ssp4_mclk"}, {.name = "ssp5_mclk"}, + {.name = "ssp0_sclk"}, {.name = "ssp1_sclk"}, {.name = "ssp2_sclk"}, + {.name = "ssp3_sclk"}, {.name = "ssp4_sclk"}, {.name = "ssp5_sclk"}, + {.name = "ssp0_sclkfs"}, {.name = "ssp1_sclkfs"}, + {.name = "ssp2_sclkfs"}, + {.name = "ssp3_sclkfs"}, {.name = "ssp4_sclkfs"}, + {.name = "ssp5_sclkfs"}, +}; + +static int skl_find_machine(struct skl *skl, void *driver_data) { - struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); - struct platform_device *pdev; struct snd_soc_acpi_mach *mach = driver_data; - int ret; + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct skl_machine_pdata *pdata; mach = snd_soc_acpi_find_machine(mach); if (mach == NULL) { dev_err(bus->dev, "No matching machine driver found\n"); return -ENODEV; } + + skl->mach = mach; skl->fw_name = mach->fw_filename; + pdata = skl->mach->pdata; + + if (mach->pdata) + skl->use_tplg_pcm = pdata->use_tplg_pcm; + + return 0; +} + +static int skl_machine_device_register(struct skl *skl) +{ + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct snd_soc_acpi_mach *mach = skl->mach; + struct platform_device *pdev; + int ret; pdev = platform_device_alloc(mach->drv_name, -1); if (pdev == NULL) { @@ -462,11 +495,8 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) return -EIO; } - if (mach->pdata) { - skl->use_tplg_pcm = - ((struct skl_machine_pdata *)mach->pdata)->use_tplg_pcm; + if (mach->pdata) dev_set_drvdata(&pdev->dev, mach->pdata); - } skl->i2s_dev = pdev; @@ -509,6 +539,74 @@ static void skl_dmic_device_unregister(struct skl *skl) platform_device_unregister(skl->dmic_dev); } +static struct skl_clk_parent_src skl_clk_src[] = { + { .clk_id = SKL_XTAL, .name = "xtal" }, + { .clk_id = SKL_CARDINAL, .name = "cardinal", .rate = 24576000 }, + { .clk_id = SKL_PLL, .name = "pll", .rate = 96000000 }, +}; + +struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(skl_clk_src); i++) { + if (skl_clk_src[i].clk_id == clk_id) + return &skl_clk_src[i]; + } + + return NULL; +} + +static void init_skl_xtal_rate(int pci_id) +{ + switch (pci_id) { + case 0x9d70: + case 0x9d71: + skl_clk_src[0].rate = 24000000; + return; + + default: + skl_clk_src[0].rate = 19200000; + return; + } +} + +static int skl_clock_device_register(struct skl *skl) +{ + struct platform_device_info pdevinfo = {NULL}; + struct skl_clk_pdata *clk_pdata; + + clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), + GFP_KERNEL); + if (!clk_pdata) + return -ENOMEM; + + init_skl_xtal_rate(skl->pci->device); + + clk_pdata->parent_clks = skl_clk_src; + clk_pdata->ssp_clks = skl_ssp_clks; + clk_pdata->num_clks = ARRAY_SIZE(skl_ssp_clks); + + /* Query NHLT to fill the rates and parent */ + skl_get_clks(skl, clk_pdata->ssp_clks); + clk_pdata->pvt_data = skl; + + /* Register Platform device */ + pdevinfo.parent = &skl->pci->dev; + pdevinfo.id = -1; + pdevinfo.name = "skl-ssp-clk"; + pdevinfo.data = clk_pdata; + pdevinfo.size_data = sizeof(*clk_pdata); + skl->clk_dev = platform_device_register_full(&pdevinfo); + return PTR_ERR_OR_ZERO(skl->clk_dev); +} + +static void skl_clock_device_unregister(struct skl *skl) +{ + if (skl->clk_dev) + platform_device_unregister(skl->clk_dev); +} + /* * Probe the given codec address */ @@ -615,18 +713,30 @@ static void skl_probe_work(struct work_struct *work) /* create codec instances */ skl_codec_create(ebus); + /* register platform dai and controls */ + err = skl_platform_register(bus->dev); + if (err < 0) { + dev_err(bus->dev, "platform register failed: %d\n", err); + return; + } + + if (bus->ppcap) { + err = skl_machine_device_register(skl); + if (err < 0) { + dev_err(bus->dev, "machine register failed: %d\n", err); + goto out_err; + } + } + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn off display power on i915\n"); + skl_machine_device_unregister(skl); return; } } - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) - return; /* * we are done probing so decrement link counts */ @@ -791,18 +901,21 @@ static int skl_probe(struct pci_dev *pci, /* check if dsp is there */ if (bus->ppcap) { - err = skl_machine_device_register(skl, - (void *)pci_id->driver_data); + /* create device for dsp clk */ + err = skl_clock_device_register(skl); + if (err < 0) + goto out_clk_free; + + err = skl_find_machine(skl, (void *)pci_id->driver_data); if (err < 0) goto out_nhlt_free; err = skl_init_dsp(skl); if (err < 0) { dev_dbg(bus->dev, "error failed to register dsp\n"); - goto out_mach_free; + goto out_nhlt_free; } skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; - } if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); @@ -820,8 +933,8 @@ static int skl_probe(struct pci_dev *pci, out_dsp_free: skl_free_dsp(skl); -out_mach_free: - skl_machine_device_unregister(skl); +out_clk_free: + skl_clock_device_unregister(skl); out_nhlt_free: skl_nhlt_free(skl->nhlt); out_free: @@ -872,6 +985,7 @@ static void skl_remove(struct pci_dev *pci) skl_free_dsp(skl); skl_machine_device_unregister(skl); skl_dmic_device_unregister(skl); + skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); skl_nhlt_free(skl->nhlt); skl_free(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index e00cde8200dd..f411579bc713 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -25,9 +25,12 @@ #include <sound/hdaudio_ext.h> #include <sound/soc.h> #include "skl-nhlt.h" +#include "skl-ssp-clk.h" #define SKL_SUSPEND_DELAY 2000 +#define SKL_MAX_ASTATE_CFG 3 + #define AZX_PCIREG_PGCTL 0x44 #define AZX_PGCTL_LSRMD_MASK (1 << 4) #define AZX_PCIREG_CGCTL 0x48 @@ -45,6 +48,20 @@ struct skl_dsp_resource { struct skl_debug; +struct skl_astate_param { + u32 kcps; + u32 clk_src; +}; + +struct skl_astate_config { + u32 count; + struct skl_astate_param astate_table[0]; +}; + +struct skl_fw_config { + struct skl_astate_config *astate_cfg; +}; + struct skl { struct hdac_ext_bus ebus; struct pci_dev *pci; @@ -52,6 +69,7 @@ struct skl { unsigned int init_done:1; /* delayed init status */ struct platform_device *dmic_dev; struct platform_device *i2s_dev; + struct platform_device *clk_dev; struct snd_soc_platform *platform; struct snd_soc_dai_driver *dais; @@ -75,6 +93,8 @@ struct skl { u8 nr_modules; struct skl_module **modules; bool use_tplg_pcm; + struct skl_fw_config cfg; + struct snd_soc_acpi_mach *mach; }; #define skl_to_ebus(s) (&(s)->ebus) @@ -125,6 +145,8 @@ const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id); void skl_update_d0i3c(struct device *dev, bool enable); int skl_nhlt_create_sysfs(struct skl *skl); void skl_nhlt_remove_sysfs(struct skl *skl); +void skl_get_clks(struct skl *skl, struct skl_ssp_clk *ssp_clks); +struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id); struct skl_module_cfg; diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 8fda182f849b..a7362d1cda1b 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1590,12 +1590,16 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, afe); - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) - goto err_pm_disable; - pm_runtime_get_sync(&pdev->dev); - ret = snd_soc_register_platform(&pdev->dev, &mtk_afe_pcm_platform); + pm_runtime_enable(dev); + if (!pm_runtime_enabled(dev)) { + ret = mt2701_afe_runtime_resume(dev); + if (ret) + goto err_pm_disable; + } + pm_runtime_get_sync(dev); + + ret = snd_soc_register_platform(dev, &mtk_afe_pcm_platform); if (ret) { dev_warn(dev, "err_platform\n"); goto err_platform; @@ -1610,35 +1614,28 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) goto err_dai_component; } - mt2701_afe_runtime_resume(&pdev->dev); - return 0; err_dai_component: - snd_soc_unregister_component(&pdev->dev); - + snd_soc_unregister_platform(dev); err_platform: - snd_soc_unregister_platform(&pdev->dev); - + pm_runtime_put_sync(dev); err_pm_disable: - pm_runtime_disable(&pdev->dev); + pm_runtime_disable(dev); return ret; } static int mt2701_afe_pcm_dev_remove(struct platform_device *pdev) { - struct mtk_base_afe *afe = platform_get_drvdata(pdev); - + pm_runtime_put_sync(&pdev->dev); pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt2701_afe_runtime_suspend(&pdev->dev); - pm_runtime_put_sync(&pdev->dev); snd_soc_unregister_component(&pdev->dev); snd_soc_unregister_platform(&pdev->dev); - /* disable afe clock */ - mt2701_afe_disable_clock(afe); + return 0; } diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index d40219678700..cb72c1e57da0 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -105,7 +105,7 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, int pin, changed = 0; /* Refuse any mode changes if we are not able to control the codec. */ - if (!cx20442_codec->hw_write) + if (!cx20442_codec->component.card->pop_time) return -EUNATCH; if (ucontrol->value.enumerated.item[0] >= control->items) @@ -345,7 +345,7 @@ static void cx81801_receive(struct tty_struct *tty, if (!codec) return; - if (!codec->hw_write) { + if (!codec->component.card->pop_time) { /* First modem response, complete setup procedure */ /* Initialize timer used for config pulse generation */ diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index d49adc822a11..704428735e3c 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -43,7 +43,7 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec; + struct snd_soc_component *component; struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); @@ -92,7 +92,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) jack = pdata->jack.jack; - snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); @@ -102,15 +102,15 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) for (i = 0 ; i < dai_link->num_codecs; i++) { struct snd_soc_dai *dai = rtd->codec_dais[i]; - codec = dai->codec; + component = dai->component; /* Set default mclk for internal codec */ - rval = snd_soc_codec_set_sysclk(codec, 0, 0, DEFAULT_MCLK_RATE, + rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE, SND_SOC_CLOCK_IN); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set mclk: %d\n", rval); return rval; } - rval = snd_soc_codec_set_jack(codec, &pdata->jack, NULL); + rval = snd_soc_component_set_jack(component, &pdata->jack, NULL); if (rval != 0 && rval != -ENOTSUPP) { dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index d64fbbd50544..fa6cd1de828b 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -206,7 +206,8 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key( + rockchip_sound_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key( rockchip_sound_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key( diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 908211e1d6fc..950823d69e9c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -328,6 +328,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, val |= I2S_CHN_4; break; case 2: + case 1: val |= I2S_CHN_2; break; default: @@ -460,7 +461,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = { }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = (SNDRV_PCM_FMTBIT_S8 | @@ -504,6 +505,7 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_INTCR: case I2S_XFER: case I2S_CLR: + case I2S_TXDR: case I2S_RXDR: case I2S_FIFOLR: case I2S_INTSR: @@ -518,6 +520,9 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case I2S_INTSR: case I2S_CLR: + case I2S_FIFOLR: + case I2S_TXDR: + case I2S_RXDR: return true; default: return false; @@ -527,6 +532,8 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { + case I2S_RXDR: + return true; default: return false; } @@ -654,7 +661,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) { - if (val >= 2 && val <= 8) + if (val >= 1 && val <= 8) soc_dai->capture.channels_max = val; } diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index ee5055d47d13..a89fe9b6463b 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -322,26 +322,30 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); if (IS_ERR(spdif->mclk)) { dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - return PTR_ERR(spdif->mclk); + ret = PTR_ERR(spdif->mclk); + goto err_disable_hclk; } ret = clk_prepare_enable(spdif->mclk); if (ret) { dev_err(spdif->dev, "clock enable failed %d\n", ret); - return ret; + goto err_disable_clocks; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_disable_clocks; + } spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(spdif->regmap); + ret = PTR_ERR(spdif->regmap); + goto err_disable_clocks; } spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; @@ -373,6 +377,10 @@ static int rk_spdif_probe(struct platform_device *pdev) err_pm_runtime: pm_runtime_disable(&pdev->dev); +err_disable_clocks: + clk_disable_unprepare(spdif->mclk); +err_disable_hclk: + clk_disable_unprepare(spdif->hclk); return ret; } diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 8ddb08714faa..4672688cac32 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -222,7 +222,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod, NULL, &val, NULL); val = val << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; rsnd_mod_bset(adg_mod, CMDOUT_TIMSEL, mask, val); @@ -250,7 +250,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, in = in << shift; out = out << shift; - mask = 0xffff << shift; + mask = 0x0f1f << shift; switch (id / 2) { case 0: @@ -380,7 +380,7 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) ckr = 0x80000000; } - rsnd_mod_bset(adg_mod, BRGCKR, 0x80FF0000, adg->ckr | ckr); + rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr | ckr); rsnd_mod_write(adg_mod, BRRA, adg->rbga); rsnd_mod_write(adg_mod, BRRB, adg->rbgb); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index c70eb2097816..64d5ecb86528 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,16 +197,27 @@ int rsnd_io_is_working(struct rsnd_dai_stream *io) return 0; } -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - return runtime->channels; + /* + * params will be added when refine + * see + * __rsnd_soc_hw_rule_rate() + * __rsnd_soc_hw_rule_channels() + */ + if (params) + return params_channels(params); + else + return runtime->channels; } -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { - int chan = rsnd_runtime_channel_original(io); + int chan = rsnd_runtime_channel_original_with_params(io, params); struct rsnd_mod *ctu_mod = rsnd_io_to_mod_ctu(io); if (ctu_mod) { @@ -219,12 +230,13 @@ int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io) return chan; } -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); int chan = rsnd_io_is_play(io) ? - rsnd_runtime_channel_after_ctu(io) : - rsnd_runtime_channel_original(io); + rsnd_runtime_channel_after_ctu_with_params(io, params) : + rsnd_runtime_channel_original_with_params(io, params); /* Use Multi SSI */ if (rsnd_runtime_is_ssi_multi(io)) @@ -262,10 +274,10 @@ u32 rsnd_get_adinr_bit(struct rsnd_mod *mod, struct rsnd_dai_stream *io) struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct device *dev = rsnd_priv_to_dev(priv); - switch (runtime->sample_bits) { + switch (snd_pcm_format_width(runtime->format)) { case 16: return 8 << 16; - case 32: + case 24: return 0 << 16; } @@ -282,11 +294,12 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io); struct rsnd_mod *target; struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - u32 val = 0x76543210; - u32 mask = ~0; /* - * *Hardware* L/R and *Software* L/R are inverted. + * *Hardware* L/R and *Software* L/R are inverted for 16bit data. + * 31..16 15...0 + * HW: [L ch] [R ch] + * SW: [R ch] [L ch] * We need to care about inversion timing to control * Playback/Capture correctly. * The point is [DVC] needs *Hardware* L/R, [MEM] needs *Software* L/R @@ -313,27 +326,13 @@ u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io) target = cmd ? cmd : ssiu; } - mask <<= runtime->channels * 4; - val = val & mask; - - switch (runtime->sample_bits) { - case 16: - val |= 0x67452301 & ~mask; - break; - case 32: - val |= 0x76543210 & ~mask; - break; - } - - /* - * exchange channeles on SRC if possible, - * otherwise, R/L volume settings on DVC - * changes inverted channels - */ - if (mod == target) - return val; - else + /* Non target mod or 24bit data needs normal DALIGN */ + if ((snd_pcm_format_width(runtime->format) != 16) || + (mod != target)) return 0x76543210; + /* Target mod needs inverted DALIGN when 16bit */ + else + return 0x67452301; } u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) @@ -363,12 +362,8 @@ u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod) * HW 24bit data is located as 0x******00 * */ - switch (runtime->sample_bits) { - case 16: + if (snd_pcm_format_width(runtime->format) == 16) return 0; - case 32: - break; - } for (i = 0; i < ARRAY_SIZE(playback_mods); i++) { tmod = rsnd_io_to_mod(io, mods[i]); @@ -616,8 +611,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - rsnd_dai_stream_init(io, substream); - ret = rsnd_dai_call(init, io, priv); if (ret < 0) goto dai_trigger_end; @@ -639,7 +632,6 @@ static int rsnd_soc_dai_trigger(struct snd_pcm_substream *substream, int cmd, ret |= rsnd_dai_call(quit, io, priv); - rsnd_dai_stream_quit(io); break; default: ret = -EINVAL; @@ -784,8 +776,9 @@ static int rsnd_soc_hw_rule(struct rsnd_priv *priv, return snd_interval_refine(iv, &p); } -static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -793,25 +786,37 @@ static int rsnd_soc_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_rate_list, ARRAY_SIZE(rsnd_soc_hw_rate_list), &ic, ir); } +static int rsnd_soc_hw_rule_rate_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 1); +} + +static int rsnd_soc_hw_rule_rate_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_rate(params, rule, 0); +} -static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) +static int __rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule, + int is_play) { struct snd_interval *ic_ = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_interval *ir = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); @@ -819,22 +824,34 @@ static int rsnd_soc_hw_rule_channels(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; /* * possible sampling rate limitation is same as * 2ch if it supports multi ssi + * and same as 8ch if TDM 6ch (see rsnd_ssi_config_init()) */ ic = *ic_; - if (1 < rsnd_rdai_ssi_lane_get(rdai)) { - ic.min = 2; - ic.max = 2; - } + ic.min = + ic.max = rsnd_runtime_channel_for_ssi_with_params(io, params); return rsnd_soc_hw_rule(priv, rsnd_soc_hw_channels_list, ARRAY_SIZE(rsnd_soc_hw_channels_list), ir, &ic); } +static int rsnd_soc_hw_rule_channels_playback(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 1); +} + +static int rsnd_soc_hw_rule_channels_capture(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + return __rsnd_soc_hw_rule_channels(params, rule, 0); +} + static const struct snd_pcm_hardware rsnd_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | @@ -859,6 +876,8 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, int ret; int i; + rsnd_dai_stream_init(io, substream); + /* * Channel Limitation * It depends on Platform design @@ -886,11 +905,17 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, * It depends on Clock Master Mode */ if (rsnd_rdai_is_clk_master(rdai)) { + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - rsnd_soc_hw_rule_rate, dai, + is_play ? rsnd_soc_hw_rule_rate_playback : + rsnd_soc_hw_rule_rate_capture, + dai, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - rsnd_soc_hw_rule_channels, dai, + is_play ? rsnd_soc_hw_rule_channels_playback : + rsnd_soc_hw_rule_channels_capture, + dai, SNDRV_PCM_HW_PARAM_RATE, -1); } @@ -915,6 +940,8 @@ static void rsnd_soc_dai_shutdown(struct snd_pcm_substream *substream, * call rsnd_dai_call without spinlock */ rsnd_dai_call(nolock_stop, io, priv); + + rsnd_dai_stream_quit(io); } static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { @@ -990,7 +1017,7 @@ of_node_compatible: static void __rsnd_dai_probe(struct rsnd_priv *priv, struct device_node *dai_np, - int dai_i, int is_graph) + int dai_i) { struct device_node *playback, *capture; struct rsnd_dai_stream *io_playback; @@ -1089,13 +1116,13 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) dai_i = 0; if (is_graph) { for_each_endpoint_of_node(dai_node, dai_np) { - __rsnd_dai_probe(priv, dai_np, dai_i, is_graph); + __rsnd_dai_probe(priv, dai_np, dai_i); rsnd_ssi_parse_hdmi_connection(priv, dai_np, dai_i); dai_i++; } } else { for_each_child_of_node(dai_node, dai_np) - __rsnd_dai_probe(priv, dai_np, dai_i++, is_graph); + __rsnd_dai_probe(priv, dai_np, dai_i++); } return 0; @@ -1332,8 +1359,8 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } @@ -1496,6 +1523,8 @@ static int rsnd_remove(struct platform_device *pdev) }; int ret = 0, i; + snd_soc_disconnect_sync(&pdev->dev); + pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fd557abfe390..41de23417c4a 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -26,10 +26,7 @@ struct rsnd_dmaen { struct dma_chan *chan; dma_cookie_t cookie; - dma_addr_t dma_buf; unsigned int dma_len; - unsigned int dma_period; - unsigned int dma_cnt; }; struct rsnd_dmapp { @@ -71,69 +68,10 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -#define rsnd_dmaen_sync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 1) -#define rsnd_dmaen_unsync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 0) -static void __rsnd_dmaen_sync(struct rsnd_dmaen *dmaen, struct rsnd_dai_stream *io, - int i, int sync) -{ - struct device *dev = dmaen->chan->device->dev; - enum dma_data_direction dir; - int is_play = rsnd_io_is_play(io); - dma_addr_t buf; - int len, max; - size_t period; - - len = dmaen->dma_len; - period = dmaen->dma_period; - max = len / period; - i = i % max; - buf = dmaen->dma_buf + (period * i); - - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - - if (sync) - dma_sync_single_for_device(dev, buf, period, dir); - else - dma_sync_single_for_cpu(dev, buf, period, dir); -} - static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { - struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - bool elapsed = false; - unsigned long flags; - - /* - * Renesas sound Gen1 needs 1 DMAC, - * Gen2 needs 2 DMAC. - * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri. - * But, Audio-DMAC-peri-peri doesn't have interrupt, - * and this driver is assuming that here. - */ - spin_lock_irqsave(&priv->lock, flags); - - if (rsnd_io_is_working(io)) { - rsnd_dmaen_unsync(dmaen, io, dmaen->dma_cnt); - - /* - * Next period is already started. - * Let's sync Next Next period - * see - * rsnd_dmaen_start() - */ - rsnd_dmaen_sync(dmaen, io, dmaen->dma_cnt + 2); - - elapsed = true; - - dmaen->dma_cnt++; - } - - spin_unlock_irqrestore(&priv->lock, flags); - - if (elapsed) + if (rsnd_io_is_working(io)) rsnd_dai_period_elapsed(io); } @@ -165,14 +103,8 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - if (dmaen->chan) { - int is_play = rsnd_io_is_play(io); - + if (dmaen->chan) dmaengine_terminate_all(dmaen->chan); - dma_unmap_single(dmaen->chan->device->dev, - dmaen->dma_buf, dmaen->dma_len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - } return 0; } @@ -237,11 +169,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; - dma_addr_t buf; - size_t len; - size_t period; int is_play = rsnd_io_is_play(io); - int i; int ret; cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; @@ -258,19 +186,10 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - len = snd_pcm_lib_buffer_bytes(substream); - period = snd_pcm_lib_period_bytes(substream); - buf = dma_map_single(dmaen->chan->device->dev, - substream->runtime->dma_area, - len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - if (dma_mapping_error(dmaen->chan->device->dev, buf)) { - dev_err(dev, "dma map failed\n"); - return -EIO; - } - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - buf, len, period, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); @@ -282,18 +201,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, desc->callback = rsnd_dmaen_complete; desc->callback_param = rsnd_mod_get(dma); - dmaen->dma_buf = buf; - dmaen->dma_len = len; - dmaen->dma_period = period; - dmaen->dma_cnt = 0; - - /* - * synchronize this and next period - * see - * __rsnd_dmaen_complete() - */ - for (i = 0; i < 2; i++) - rsnd_dmaen_sync(dmaen, io, i); + dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); dmaen->cookie = dmaengine_submit(desc); if (dmaen->cookie < 0) { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 57cd2bc773c2..ad6523595b0a 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -399,9 +399,18 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); -int rsnd_runtime_channel_original(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_after_ctu(struct rsnd_dai_stream *io); -int rsnd_runtime_channel_for_ssi(struct rsnd_dai_stream *io); +#define rsnd_runtime_channel_original(io) \ + rsnd_runtime_channel_original_with_params(io, NULL) +int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_after_ctu(io) \ + rsnd_runtime_channel_after_ctu_with_params(io, NULL) +int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); +#define rsnd_runtime_channel_for_ssi(io) \ + rsnd_runtime_channel_for_ssi_with_params(io, NULL) +int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, + struct snd_pcm_hw_params *params); int rsnd_runtime_is_ssi_multi(struct rsnd_dai_stream *io); int rsnd_runtime_is_ssi_tdm(struct rsnd_dai_stream *io); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fece1e5f582f..97a9db892a8f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -79,8 +79,8 @@ struct rsnd_ssi { int irq; unsigned int usrcnt; + /* for PIO */ int byte_pos; - int period_pos; int byte_per_period; int next_period_byte; }; @@ -371,11 +371,11 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rsnd_io_is_play(io)) cr_own |= TRMD; - switch (runtime->sample_bits) { + switch (snd_pcm_format_width(runtime->format)) { case 16: cr_own |= DWL_16; break; - case 32: + case 24: cr_own |= DWL_24; break; } @@ -414,59 +414,6 @@ static void rsnd_ssi_register_setup(struct rsnd_mod *mod) ssi->cr_en); } -static void rsnd_ssi_pointer_init(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - - ssi->byte_pos = 0; - ssi->period_pos = 0; - ssi->byte_per_period = runtime->period_size * - runtime->channels * - samples_to_bytes(runtime, 1); - ssi->next_period_byte = ssi->byte_per_period; -} - -static int rsnd_ssi_pointer_offset(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - int additional) -{ - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - int pos = ssi->byte_pos + additional; - - pos %= (runtime->periods * ssi->byte_per_period); - - return pos; -} - -static bool rsnd_ssi_pointer_update(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - int byte) -{ - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); - - ssi->byte_pos += byte; - - if (ssi->byte_pos >= ssi->next_period_byte) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - - ssi->period_pos++; - ssi->next_period_byte += ssi->byte_per_period; - - if (ssi->period_pos >= runtime->periods) { - ssi->byte_pos = 0; - ssi->period_pos = 0; - ssi->next_period_byte = ssi->byte_per_period; - } - - return true; - } - - return false; -} - /* * SSI mod common functions */ @@ -480,8 +427,6 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, if (!rsnd_ssi_is_run_mods(mod, io)) return 0; - rsnd_ssi_pointer_init(mod, io); - ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -652,6 +597,8 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, return 0; } +static bool rsnd_ssi_pio_interrupt(struct rsnd_mod *mod, + struct rsnd_dai_stream *io); static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { @@ -670,30 +617,8 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, status = rsnd_ssi_status_get(mod); /* PIO only */ - if (!is_dma && (status & DIRQ)) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - u32 *buf = (u32 *)(runtime->dma_area + - rsnd_ssi_pointer_offset(mod, io, 0)); - int shift = 0; - - switch (runtime->sample_bits) { - case 32: - shift = 8; - break; - } - - /* - * 8/16/32 data can be assesse to TDR/RDR register - * directly as 32bit data - * see rsnd_ssi_init() - */ - if (rsnd_io_is_play(io)) - rsnd_mod_write(mod, SSITDR, (*buf) << shift); - else - *buf = (rsnd_mod_read(mod, SSIRDR) >> shift); - - elapsed = rsnd_ssi_pointer_update(mod, io, sizeof(*buf)); - } + if (!is_dma && (status & DIRQ)) + elapsed = rsnd_ssi_pio_interrupt(mod, io); /* DMA only */ if (is_dma && (status & (UIRQ | OIRQ))) @@ -831,14 +756,78 @@ static int rsnd_ssi_common_remove(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_pointer(struct rsnd_mod *mod, +/* + * SSI PIO functions + */ +static bool rsnd_ssi_pio_interrupt(struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + u32 *buf = (u32 *)(runtime->dma_area + ssi->byte_pos); + int shift = 0; + int byte_pos; + bool elapsed = false; + + if (snd_pcm_format_width(runtime->format) == 24) + shift = 8; + + /* + * 8/16/32 data can be assesse to TDR/RDR register + * directly as 32bit data + * see rsnd_ssi_init() + */ + if (rsnd_io_is_play(io)) + rsnd_mod_write(mod, SSITDR, (*buf) << shift); + else + *buf = (rsnd_mod_read(mod, SSIRDR) >> shift); + + byte_pos = ssi->byte_pos + sizeof(*buf); + + if (byte_pos >= ssi->next_period_byte) { + int period_pos = byte_pos / ssi->byte_per_period; + + if (period_pos >= runtime->periods) { + byte_pos = 0; + period_pos = 0; + } + + ssi->next_period_byte = (period_pos + 1) * ssi->byte_per_period; + + elapsed = true; + } + + WRITE_ONCE(ssi->byte_pos, byte_pos); + + return elapsed; +} + +static int rsnd_ssi_pio_init(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct rsnd_priv *priv) +{ + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + + if (!rsnd_ssi_is_parent(mod, io)) { + ssi->byte_pos = 0; + ssi->byte_per_period = runtime->period_size * + runtime->channels * + samples_to_bytes(runtime, 1); + ssi->next_period_byte = ssi->byte_per_period; + } + + return rsnd_ssi_init(mod, io, priv); +} + +static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - *pointer = bytes_to_frames(runtime, ssi->byte_pos); + *pointer = bytes_to_frames(runtime, READ_ONCE(ssi->byte_pos)); return 0; } @@ -847,12 +836,12 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, .remove = rsnd_ssi_common_remove, - .init = rsnd_ssi_init, + .init = rsnd_ssi_pio_init, .quit = rsnd_ssi_quit, .start = rsnd_ssi_start, .stop = rsnd_ssi_stop, .irq = rsnd_ssi_irq, - .pointer= rsnd_ssi_pointer, + .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, }; diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d948757d300..6ff8a36c2c82 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -125,6 +125,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, { int hdmi = rsnd_ssi_hdmi_port(io); int ret; + u32 mode = 0; ret = rsnd_ssiu_init(mod, io, priv); if (ret < 0) @@ -136,9 +137,11 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, * see * rsnd_ssi_config_init() */ - rsnd_mod_write(mod, SSI_MODE, 0x1); + mode = 0x1; } + rsnd_mod_write(mod, SSI_MODE, mode); + if (rsnd_ssi_use_busif(io)) { rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_get_adinr_bit(mod, io) | diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index f21df28bc28e..3d7e1ff79139 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -16,79 +16,16 @@ #include <sound/soc-acpi.h> -static acpi_status snd_soc_acpi_find_name(acpi_handle handle, u32 level, - void *context, void **ret) -{ - struct acpi_device *adev; - const char *name = NULL; - - if (acpi_bus_get_device(handle, &adev)) - return AE_OK; - - if (adev->status.present && adev->status.functional) { - name = acpi_dev_name(adev); - *(const char **)ret = name; - return AE_CTRL_TERMINATE; - } - - return AE_OK; -} - -const char *snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) -{ - const char *name = NULL; - acpi_status status; - - status = acpi_get_devices(hid, snd_soc_acpi_find_name, NULL, - (void **)&name); - - if (ACPI_FAILURE(status) || name[0] == '\0') - return NULL; - - return name; -} -EXPORT_SYMBOL_GPL(snd_soc_acpi_find_name_from_hid); - -static acpi_status snd_soc_acpi_mach_match(acpi_handle handle, u32 level, - void *context, void **ret) -{ - unsigned long long sta; - acpi_status status; - - *(bool *)context = true; - status = acpi_evaluate_integer(handle, "_STA", NULL, &sta); - if (ACPI_FAILURE(status) || !(sta & ACPI_STA_DEVICE_PRESENT)) - *(bool *)context = false; - - return AE_OK; -} - -bool snd_soc_acpi_check_hid(const u8 hid[ACPI_ID_LEN]) -{ - acpi_status status; - bool found = false; - - status = acpi_get_devices(hid, snd_soc_acpi_mach_match, &found, NULL); - - if (ACPI_FAILURE(status)) - return false; - - return found; -} -EXPORT_SYMBOL_GPL(snd_soc_acpi_check_hid); - struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { struct snd_soc_acpi_mach *mach; for (mach = machines; mach->id[0]; mach++) { - if (snd_soc_acpi_check_hid(mach->id) == true) { - if (mach->machine_quirk == NULL) - return mach; - - if (mach->machine_quirk(mach) != NULL) - return mach; + if (acpi_dev_present(mach->id, NULL, -1)) { + if (mach->machine_quirk) + mach = mach->machine_quirk(mach); + return mach; } } return NULL; @@ -163,7 +100,7 @@ struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) return mach; for (i = 0; i < codec_list->num_codecs; i++) { - if (snd_soc_acpi_check_hid(codec_list->codecs[i]) != true) + if (!acpi_dev_present(codec_list->codecs[i], NULL, -1)) return NULL; } diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d9b1e6417fb9..81232f4ab614 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -1096,7 +1096,6 @@ static struct snd_compr_ops soc_compr_dyn_ops = { */ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; @@ -1199,8 +1198,9 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_compress_new(rtd->card->snd_card, num, direction, new_name, compr); if (ret < 0) { + component = rtd->codec_dai->component; pr_err("compress asoc: can't create compress for codec %s\n", - codec->component.name); + component->name); goto compr_err; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a13fbcba23f..d3a4b58d6b7b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -213,7 +213,7 @@ static umode_t soc_dev_attr_is_visible(struct kobject *kobj, if (attr == &dev_attr_pmdown_time.attr) return attr->mode; /* always visible */ - return rtd->codec ? attr->mode : 0; /* enabled only with codec */ + return rtd->num_codecs ? attr->mode : 0; /* enabled only with codec */ } static const struct attribute_group soc_dapm_dev_group = { @@ -598,6 +598,7 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_rtdcom_lookup); struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream) @@ -1392,6 +1393,16 @@ static int soc_init_dai_link(struct snd_soc_card *card, return 0; } +void snd_soc_disconnect_sync(struct device *dev) +{ + struct snd_soc_component *component = snd_soc_lookup_component(dev, NULL); + + if (!component || !component->card) + return; + + snd_card_disconnect_sync(component->card->snd_card); +} + /** * snd_soc_add_dai_link - Add a DAI link dynamically * @card: The ASoC card to which the DAI link is added @@ -1945,7 +1956,9 @@ int snd_soc_runtime_set_dai_fmt(struct snd_soc_pcm_runtime *rtd, } /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ - if (cpu_dai->codec) { + /* the component which has non_legacy_dai_naming is Codec */ + if (cpu_dai->codec || + cpu_dai->component->driver->non_legacy_dai_naming) { unsigned int inv_dai_fmt; inv_dai_fmt = dai_fmt & ~SND_SOC_DAIFMT_MASTER_MASK; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 20340ade20a7..2bc1c4c17896 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -34,6 +34,10 @@ int snd_soc_component_read(struct snd_soc_component *component, ret = regmap_read(component->regmap, reg, val); else if (component->read) ret = component->read(component, reg, val); + else if (component->driver->read) { + *val = component->driver->read(component, reg); + ret = 0; + } else ret = -EIO; @@ -70,6 +74,8 @@ int snd_soc_component_write(struct snd_soc_component *component, return regmap_write(component->regmap, reg, val); else if (component->write) return component->write(component, reg, val); + else if (component->driver->write) + return component->driver->write(component, reg, val); else return -EIO; } diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c730b9..7144a51ddfa9 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -378,7 +378,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1 << (fls(min + max) - 1)) - 1; unsigned int val; int ret; @@ -423,7 +423,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, unsigned int rshift = mc->rshift; int max = mc->max; int min = mc->min; - int mask = (1 << (fls(min + max) - 1)) - 1; + unsigned int mask = (1 << (fls(min + max) - 1)) - 1; int err = 0; unsigned int val, val_mask, val2 = 0; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 0537c6322990..2b4ceda36291 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -204,6 +204,10 @@ static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); + + if (len < 0) + return 0; + buf[len] = 0; return len; } @@ -1476,9 +1480,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, return -EINVAL; } csize = hdr->bControlSize; - if (csize <= 1) { + if (!csize) { usb_audio_dbg(state->chip, - "unit %u: invalid bControlSize <= 1\n", + "unit %u: invalid bControlSize == 0\n", unitid); return -EINVAL; } @@ -2169,19 +2173,25 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; - nameid = uac_selector_unit_iSelector(desc); + /* check the static mapping table at first */ len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); - if (len) - ; - else if (nameid) - snd_usb_copy_string_desc(state, nameid, kctl->id.name, - sizeof(kctl->id.name)); - else { - len = get_term_name(state, &state->oterm, + if (!len) { + /* no mapping ? */ + /* if iSelector is given, use it */ + nameid = uac_selector_unit_iSelector(desc); + if (nameid) + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, + sizeof(kctl->id.name)); + /* ... or pick up the terminal name at next */ + if (!len) + len = get_term_name(state, &state->oterm, kctl->id.name, sizeof(kctl->id.name), 0); + /* ... or use the fixed string "USB" as the last resort */ if (!len) strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name)); + /* and add the proper suffix */ if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) append_ctl_name(kctl, " Clock Source"); else if ((state->oterm.type & 0xff00) == 0x0100) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 77eecaa4db1f..a66ef5777887 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1166,10 +1166,11 @@ static bool is_marantz_denon_dac(unsigned int id) /* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch * between PCM/DOP and native DSD mode */ -static bool is_teac_50X_dac(unsigned int id) +static bool is_teac_dsd_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ return true; } return false; @@ -1202,7 +1203,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_50X_dac(subs->stream->chip->usb_id)) { + } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1392,7 +1393,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* TEAC devices with USB DAC functionality */ - if (is_teac_50X_dac(chip->usb_id)) { + if (is_teac_dsd_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } |