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| author | Takashi Iwai <tiwai@suse.de> | 2026-04-13 07:18:36 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2026-04-13 07:18:36 +0200 |
| commit | f365e47bfbe388b2dde411f8a016065274eee02f (patch) | |
| tree | 9b41ea2f46dc9d5646d807e02297518826b6fd77 | |
| parent | 52521e8398839105ef8eb22b3f0993f9b0d11a57 (diff) | |
| parent | 713e0f011178a2896e46db3244093454708066e2 (diff) | |
| download | linux-f365e47bfbe388b2dde411f8a016065274eee02f.tar.gz linux-f365e47bfbe388b2dde411f8a016065274eee02f.zip | |
Merge branch 'for-next' into for-linus
Signed-off-by: Takashi Iwai <tiwai@suse.de>
88 files changed, 2333 insertions, 687 deletions
diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 55b845d38236..f75f08763941 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -2376,6 +2376,13 @@ quirk_flags Skip the probe-time interface setup (usb_set_interface, init_pitch, init_sample_rate); redundant with snd_usb_endpoint_prepare() at stream-open time + * bit 27: ``mixer_playback_linear_vol`` + Set linear volume mapping for devices where the playback volume + control value is mapped to voltage (instead of dB) level linearly. + In short: ``x(raw) = (raw - raw_min) / (raw_max - raw_min)``; + ``V(x) = k * x``; ``dB(x) = 20 * log10(x)``. Overrides bit 24 + * bit 28: ``mixer_capture_linear_vol`` + Similar to bit 27 but for capture streams. Overrides bit 25 This module supports multiple devices, autoprobe and hotplugging. diff --git a/include/sound/core.h b/include/sound/core.h index 64327e971122..4093ec82a0a1 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -133,6 +133,9 @@ struct snd_card { #ifdef CONFIG_SND_DEBUG struct dentry *debugfs_root; /* debugfs root for card */ #endif +#ifdef CONFIG_SND_CTL_DEBUG + struct snd_ctl_elem_value *value_buf; /* buffer for kctl->put() verification */ +#endif #ifdef CONFIG_PM unsigned int power_state; /* power state */ diff --git a/include/sound/gus.h b/include/sound/gus.h index 321ae93625eb..3feb42627de1 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -536,6 +536,7 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, struct snd_gf1_dma_block * block, int atomic, int synth); +void snd_gf1_dma_suspend(struct snd_gus_card *gus); /* gus_volume.c */ @@ -552,6 +553,8 @@ struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, i void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice); int snd_gf1_start(struct snd_gus_card * gus); int snd_gf1_stop(struct snd_gus_card * gus); +int snd_gf1_suspend(struct snd_gus_card *gus); +int snd_gf1_resume(struct snd_gus_card *gus); /* gus_mixer.c */ @@ -572,6 +575,8 @@ int snd_gus_create(struct snd_card *card, int effect, struct snd_gus_card ** rgus); int snd_gus_initialize(struct snd_gus_card * gus); +int snd_gus_suspend(struct snd_gus_card *gus); +int snd_gus_resume(struct snd_gus_card *gus); /* gus_irq.c */ @@ -583,6 +588,8 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus); /* gus_uart.c */ int snd_gf1_rawmidi_new(struct snd_gus_card *gus, int device); +void snd_gf1_uart_suspend(struct snd_gus_card *gus); +void snd_gf1_uart_resume(struct snd_gus_card *gus); /* gus_dram.c */ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr, @@ -593,5 +600,6 @@ int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr, /* gus_timer.c */ void snd_gf1_timers_init(struct snd_gus_card *gus); void snd_gf1_timers_done(struct snd_gus_card *gus); +void snd_gf1_timers_resume(struct snd_gus_card *gus); #endif /* __SOUND_GUS_H */ diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 5d9f0ef228af..24581080e26a 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -336,6 +336,17 @@ snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags, return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm); } +/* sync after write */ +static inline int +snd_hda_codec_write_sync(struct hda_codec *codec, hda_nid_t nid, int flags, + unsigned int verb, unsigned int parm) +{ + /* use snd_hda_codec_read() for writing; + * the returned value is usually discarded + */ + return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm); +} + #define snd_hda_param_read(codec, nid, param) \ snd_hdac_read_parm(&(codec)->core, nid, param) #define snd_hda_get_sub_nodes(codec, nid, start_nid) \ @@ -470,6 +481,10 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +void snd_hda_codec_set_gpio(struct hda_codec *codec, unsigned int mask, + unsigned int dir, unsigned int data, + unsigned int delay); + int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); /* diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index 006d358acce2..6066954409aa 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -56,7 +56,12 @@ enum { #define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d #define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e /* unused */ #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f -/* f10-f1a: GPIO */ +/* f10-f1a: GPI/GPO/GPIO */ +#define AC_VERB_GET_GPI_DATA 0x0f10 +#define AC_VERB_GET_GPI_WAKE_MASK 0x0f11 +#define AC_VERB_GET_GPI_UNSOLICITED_RSP_MASK 0x0f12 +#define AC_VERB_GET_GPI_STICKY_MASK 0x0f13 +#define AC_VERB_GET_GPO_DATA 0x0f14 #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 @@ -99,6 +104,11 @@ enum { #define AC_VERB_SET_DIGI_CONVERT_2 0x70e #define AC_VERB_SET_DIGI_CONVERT_3 0x73e #define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f +#define AC_VERB_SET_GPI_DATA 0x710 +#define AC_VERB_SET_GPI_WAKE_MASK 0x711 +#define AC_VERB_SET_SPI_UNSOLICITED_RSP_MASK 0x712 +#define AC_VERB_SET_GPI_STICKY_MASK 0x713 +#define AC_VERB_SET_GPO_DATA 0x714 #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a7860c047503..76fc33dce537 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -729,6 +729,10 @@ static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime, runtime->status->state = state; /* copy for mmap */ } +void snd_pcm_set_state(struct snd_pcm_substream *substream, + snd_pcm_state_t state); +snd_pcm_state_t snd_pcm_get_state(struct snd_pcm_substream *substream); + /** * bytes_to_samples - Unit conversion of the size from bytes to samples * @runtime: PCM runtime instance diff --git a/include/sound/tea6330t.h b/include/sound/tea6330t.h index 1c77b78f6533..3a34033d2aa3 100644 --- a/include/sound/tea6330t.h +++ b/include/sound/tea6330t.h @@ -12,5 +12,6 @@ int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer); int snd_tea6330t_update_mixer(struct snd_card *card, struct snd_i2c_bus *bus, int equalizer, int fader); +int snd_tea6330t_restore_mixer(struct snd_i2c_bus *bus); #endif /* __SOUND_TEA6330T_H */ diff --git a/include/sound/timer.h b/include/sound/timer.h index 760e132cc0cd..83bafe70cf33 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -102,6 +102,7 @@ struct snd_timer_instance { unsigned int slave_id; struct list_head open_list; struct list_head active_list; + struct list_head master_list; struct list_head ack_list; struct list_head slave_list_head; struct list_head slave_active_head; diff --git a/sound/aoa/aoa.h b/sound/aoa/aoa.h index badff9f7cd54..b92593f170ca 100644 --- a/sound/aoa/aoa.h +++ b/sound/aoa/aoa.h @@ -48,7 +48,7 @@ struct aoa_codec { u32 connected; /* data the fabric can associate with this structure */ - void *fabric_data; + const void *fabric_data; /* private! */ struct list_head list; diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 04961c456d2c..4fb593e88fb1 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -32,6 +32,7 @@ #include <linux/module.h> #include <linux/of.h> #include <linux/slab.h> +#include <sound/asoundef.h> MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); @@ -514,8 +515,36 @@ static int onyx_spdif_put(struct snd_kcontrol *kcontrol, return 1; } +static int onyx_set_spdif_pcm_rate(struct onyx *onyx, unsigned int rate) +{ + u8 dig_info3, fs; + + switch (rate) { + case 32000: + fs = IEC958_AES3_CON_FS_32000; + break; + case 44100: + fs = IEC958_AES3_CON_FS_44100; + break; + case 48000: + fs = IEC958_AES3_CON_FS_48000; + break; + default: + return -EINVAL; + } + + if (onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &dig_info3)) + return -EBUSY; + dig_info3 = (dig_info3 & ~IEC958_AES3_CON_FS) | fs; + if (onyx_write_register(onyx, ONYX_REG_DIG_INFO3, dig_info3)) + return -EBUSY; + + return 0; +} + static const struct snd_kcontrol_new onyx_spdif_ctrl = { - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), .info = onyx_spdif_info, @@ -695,9 +724,9 @@ static int onyx_prepare(struct codec_info_item *cii, case 32000: case 44100: case 48000: - /* these rates are ok for all outputs */ - /* FIXME: program spdif channel control bits here so that - * userspace doesn't have to if it only plays pcm! */ + if (onyx->codec.connected & 2) + return onyx_set_spdif_pcm_rate(onyx, + substream->runtime->rate); return 0; default: /* got some rate that the digital output can't do, @@ -980,10 +1009,12 @@ static int onyx_i2c_probe(struct i2c_client *client) onyx->codec.node = of_node_get(node); if (aoa_codec_register(&onyx->codec)) { - goto fail; + goto fail_put; } printk(KERN_DEBUG PFX "created and attached onyx instance\n"); return 0; + fail_put: + of_node_put(onyx->codec.node); fail: kfree(onyx); return -ENODEV; diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 13da2b159ad0..25214d3da65d 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -872,6 +872,7 @@ static int tas_i2c_probe(struct i2c_client *client) return 0; fail: mutex_destroy(&tas->mtx); + of_node_put(tas->codec.node); kfree(tas); return -EINVAL; } diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index c18b55305294..c3ebb6de4789 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -55,7 +55,7 @@ struct codec_connection { struct codec_connect_info { char *name; - struct codec_connection *connections; + const struct codec_connection *connections; }; #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) @@ -116,7 +116,7 @@ MODULE_ALIAS("aoa-device-id-35"); MODULE_ALIAS("aoa-device-id-44"); /* onyx with all but microphone connected */ -static struct codec_connection onyx_connections_nomic[] = { +static const struct codec_connection onyx_connections_nomic[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, .codec_bit = 0, @@ -133,7 +133,7 @@ static struct codec_connection onyx_connections_nomic[] = { }; /* onyx on machines without headphone */ -static struct codec_connection onyx_connections_noheadphones[] = { +static const struct codec_connection onyx_connections_noheadphones[] = { { .connected = CC_SPEAKERS | CC_LINEOUT | CC_LINEOUT_LABELLED_HEADPHONE, @@ -157,7 +157,7 @@ static struct codec_connection onyx_connections_noheadphones[] = { }; /* onyx on machines with real line-out */ -static struct codec_connection onyx_connections_reallineout[] = { +static const struct codec_connection onyx_connections_reallineout[] = { { .connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE, .codec_bit = 0, @@ -174,7 +174,7 @@ static struct codec_connection onyx_connections_reallineout[] = { }; /* tas on machines without line out */ -static struct codec_connection tas_connections_nolineout[] = { +static const struct codec_connection tas_connections_nolineout[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE, .codec_bit = 0, @@ -191,7 +191,7 @@ static struct codec_connection tas_connections_nolineout[] = { }; /* tas on machines with neither line out nor line in */ -static struct codec_connection tas_connections_noline[] = { +static const struct codec_connection tas_connections_noline[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE, .codec_bit = 0, @@ -204,7 +204,7 @@ static struct codec_connection tas_connections_noline[] = { }; /* tas on machines without microphone */ -static struct codec_connection tas_connections_nomic[] = { +static const struct codec_connection tas_connections_nomic[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, .codec_bit = 0, @@ -217,7 +217,7 @@ static struct codec_connection tas_connections_nomic[] = { }; /* tas on machines with everything connected */ -static struct codec_connection tas_connections_all[] = { +static const struct codec_connection tas_connections_all[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, .codec_bit = 0, @@ -233,7 +233,7 @@ static struct codec_connection tas_connections_all[] = { {} /* terminate array by .connected == 0 */ }; -static struct codec_connection toonie_connections[] = { +static const struct codec_connection toonie_connections[] = { { .connected = CC_SPEAKERS | CC_HEADPHONE, .codec_bit = 0, @@ -241,7 +241,7 @@ static struct codec_connection toonie_connections[] = { {} /* terminate array by .connected == 0 */ }; -static struct codec_connection topaz_input[] = { +static const struct codec_connection topaz_input[] = { { .connected = CC_DIGITALIN, .codec_bit = 0, @@ -249,7 +249,7 @@ static struct codec_connection topaz_input[] = { {} /* terminate array by .connected == 0 */ }; -static struct codec_connection topaz_output[] = { +static const struct codec_connection topaz_output[] = { { .connected = CC_DIGITALOUT, .codec_bit = 1, @@ -257,7 +257,7 @@ static struct codec_connection topaz_output[] = { {} /* terminate array by .connected == 0 */ }; -static struct codec_connection topaz_inout[] = { +static const struct codec_connection topaz_inout[] = { { .connected = CC_DIGITALIN, .codec_bit = 0, @@ -772,7 +772,7 @@ static int check_codec(struct aoa_codec *codec, { const u32 *ref; char propname[32]; - struct codec_connection *cc; + const struct codec_connection *cc; /* if the codec has a 'codec' node, we require a reference */ if (of_node_name_eq(codec->node, "codec")) { @@ -895,7 +895,7 @@ static void layout_notify(void *data) static void layout_attached_codec(struct aoa_codec *codec) { - struct codec_connection *cc; + const struct codec_connection *cc; struct snd_kcontrol *ctl; int headphones, lineout; struct layout_dev *ldev = layout_device; diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index f974b96e98cd..833c44c0a950 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -84,6 +84,7 @@ static void i2sbus_release_dev(struct device *dev) for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) free_irq(i2sdev->interrupts[i], i2sdev); i2sbus_control_remove_dev(i2sdev->control, i2sdev); + of_node_put(i2sdev->sound.ofdev.dev.of_node); mutex_destroy(&i2sdev->lock); kfree(i2sdev); } @@ -147,7 +148,6 @@ static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index, } /* Returns 1 if added, 0 for otherwise; don't return a negative value! */ -/* FIXME: look at device node refcounting */ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_control *control, struct device_node *np) @@ -178,8 +178,9 @@ static int i2sbus_add_dev(struct macio_dev *macio, i = 0; for_each_child_of_node(np, child) { if (of_node_name_eq(child, "sound")) { + of_node_put(sound); i++; - sound = child; + sound = of_node_get(child); } } if (i == 1) { @@ -205,6 +206,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, } } } + of_node_put(sound); /* for the time being, until we can handle non-layout-id * things in some fabric, refuse to attach if there is no * layout-id property or we haven't been forced to attach. @@ -219,7 +221,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, mutex_init(&dev->lock); spin_lock_init(&dev->low_lock); dev->sound.ofdev.archdata.dma_mask = macio->ofdev.archdata.dma_mask; - dev->sound.ofdev.dev.of_node = np; + dev->sound.ofdev.dev.of_node = of_node_get(np); dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.archdata.dma_mask; dev->sound.ofdev.dev.parent = &macio->ofdev.dev; dev->sound.ofdev.dev.release = i2sbus_release_dev; @@ -327,6 +329,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, for (i=0;i<3;i++) release_and_free_resource(dev->allocated_resource[i]); mutex_destroy(&dev->lock); + of_node_put(dev->sound.ofdev.dev.of_node); kfree(dev); return 0; } @@ -405,6 +408,9 @@ static int i2sbus_resume(struct macio_dev* dev) int err, ret = 0; list_for_each_entry(i2sdev, &control->list, item) { + if (list_empty(&i2sdev->sound.codec_list)) + continue; + /* reset i2s bus format etc. */ i2sbus_pcm_prepare_both(i2sdev); diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index aff99003d833..63004ece94f9 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -165,17 +165,16 @@ static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in) * currently in use (if any). */ hw->rate_min = 5512; hw->rate_max = 192000; - /* if the other stream is active, then we can only - * support what it is currently using. - * FIXME: I lied. This comment is wrong. We can support - * anything that works with the same serial format, ie. - * when recording 24 bit sound we can well play 16 bit - * sound at the same time iff using the same transfer mode. + /* If the other stream is already prepared, keep this stream + * on the same duplex format and rate. + * + * i2sbus_pcm_prepare() still programs one shared transport + * configuration for both directions, so mixed duplex formats + * are not supported here. */ if (other->active) { - /* FIXME: is this guaranteed by the alsa api? */ hw->formats &= pcm_format_to_bits(i2sdev->format); - /* see above, restrict rates to the one we already have */ + /* Restrict rates to the one already in use. */ hw->rate_min = i2sdev->rate; hw->rate_max = i2sdev->rate; } @@ -283,6 +282,23 @@ void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev) } #endif +static void i2sbus_pcm_clear_active(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi; + + guard(mutex)(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, NULL); + pi->active = 0; +} + +static inline int i2sbus_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, int in) +{ + i2sbus_pcm_clear_active(snd_pcm_substream_chip(substream), in); + return 0; +} + static inline int i2sbus_hw_free(struct snd_pcm_substream *substream, int in) { struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); @@ -291,14 +307,27 @@ static inline int i2sbus_hw_free(struct snd_pcm_substream *substream, int in) get_pcm_info(i2sdev, in, &pi, NULL); if (pi->dbdma_ring.stopping) i2sbus_wait_for_stop(i2sdev, pi); + i2sbus_pcm_clear_active(i2sdev, in); return 0; } +static int i2sbus_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + return i2sbus_hw_params(substream, params, 0); +} + static int i2sbus_playback_hw_free(struct snd_pcm_substream *substream) { return i2sbus_hw_free(substream, 0); } +static int i2sbus_record_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + return i2sbus_hw_params(substream, params, 1); +} + static int i2sbus_record_hw_free(struct snd_pcm_substream *substream) { return i2sbus_hw_free(substream, 1); @@ -335,7 +364,6 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) return -EINVAL; runtime = pi->substream->runtime; - pi->active = 1; if (other->active && ((i2sdev->format != runtime->format) || (i2sdev->rate != runtime->rate))) @@ -383,6 +411,9 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) /* set stop command */ command->command = cpu_to_le16(DBDMA_STOP); + cii = list_first_entry(&i2sdev->sound.codec_list, + struct codec_info_item, list); + /* ok, let's set the serial format and stuff */ switch (runtime->format) { /* 16 bit formats */ @@ -390,13 +421,7 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) case SNDRV_PCM_FORMAT_U16_BE: /* FIXME: if we add different bus factors we need to * do more here!! */ - bi.bus_factor = 0; - list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { - bi.bus_factor = cii->codec->bus_factor; - break; - } - if (!bi.bus_factor) - return -ENODEV; + bi.bus_factor = cii->codec->bus_factor; input_16bit = 1; break; case SNDRV_PCM_FORMAT_S32_BE: @@ -410,10 +435,7 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) return -EINVAL; } /* we assume all sysclocks are the same! */ - list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { - bi.sysclock_factor = cii->codec->sysclock_factor; - break; - } + bi.sysclock_factor = cii->codec->sysclock_factor; if (clock_and_divisors(bi.sysclock_factor, bi.bus_factor, @@ -450,9 +472,11 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) /* early exit if already programmed correctly */ /* not locking these is fine since we touch them only in this function */ - if (in_le32(&i2sdev->intfregs->serial_format) == sfr - && in_le32(&i2sdev->intfregs->data_word_sizes) == dws) + if (in_le32(&i2sdev->intfregs->serial_format) == sfr && + in_le32(&i2sdev->intfregs->data_word_sizes) == dws) { + pi->active = 1; return 0; + } /* let's notify the codecs about clocks going away. * For now we only do mastering on the i2s cell... */ @@ -490,6 +514,7 @@ static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in) if (cii->codec->switch_clock) cii->codec->switch_clock(cii, CLOCK_SWITCH_SLAVE); + pi->active = 1; return 0; } @@ -734,6 +759,7 @@ static snd_pcm_uframes_t i2sbus_playback_pointer(struct snd_pcm_substream static const struct snd_pcm_ops i2sbus_playback_ops = { .open = i2sbus_playback_open, .close = i2sbus_playback_close, + .hw_params = i2sbus_playback_hw_params, .hw_free = i2sbus_playback_hw_free, .prepare = i2sbus_playback_prepare, .trigger = i2sbus_playback_trigger, @@ -802,6 +828,7 @@ static snd_pcm_uframes_t i2sbus_record_pointer(struct snd_pcm_substream static const struct snd_pcm_ops i2sbus_record_ops = { .open = i2sbus_record_open, .close = i2sbus_record_close, + .hw_params = i2sbus_record_hw_params, .hw_free = i2sbus_record_hw_free, .prepare = i2sbus_record_prepare, .trigger = i2sbus_record_trigger, diff --git a/sound/core/Makefile b/sound/core/Makefile index 31a0623cc89d..fdd3bb6e81a9 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -23,6 +23,7 @@ snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o # for trace-points CFLAGS_pcm_lib.o := -I$(src) CFLAGS_pcm_native.o := -I$(src) +CFLAGS_control.o := -I$(src) snd-pcm-dmaengine-y := pcm_dmaengine.o diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index fdba6e4b25fd..fd63d219bf86 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -41,13 +41,6 @@ #define COMPR_CODEC_CAPS_OVERFLOW #endif -/* TODO: - * - add substream support for multiple devices in case of - * SND_DYNAMIC_MINORS is not used - * - Multiple node representation - * driver should be able to register multiple nodes - */ - struct snd_compr_file { unsigned long caps; struct snd_compr_stream stream; @@ -190,9 +183,21 @@ snd_compr_tstamp32_from_64(struct snd_compr_tstamp *tstamp32, static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp64 *tstamp) { + int ret; + if (!stream->ops->pointer) return -ENOTSUPP; - stream->ops->pointer(stream, tstamp); + + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + return -EBADFD; + default: + break; + } + + ret = stream->ops->pointer(stream, tstamp); + if (ret != 0) + return ret; pr_debug("dsp consumed till %u total %llu bytes\n", tstamp->byte_offset, tstamp->copied_total); if (stream->direction == SND_COMPRESS_PLAYBACK) diff --git a/sound/core/control.c b/sound/core/control.c index 934e84e93838..374e703d15a9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -19,6 +19,13 @@ #include <sound/info.h> #include <sound/control.h> +#ifdef CONFIG_SND_CTL_DEBUG +#define CREATE_TRACE_POINTS +#include "control_trace.h" +#else +#define trace_snd_ctl_put(card, kctl, iname, expected, actual) +#endif + // Max allocation size for user controls. static int max_user_ctl_alloc_size = 8 * 1024 * 1024; module_param_named(max_user_ctl_alloc_size, max_user_ctl_alloc_size, int, 0444); @@ -1264,6 +1271,72 @@ static int snd_ctl_elem_read_user(struct snd_card *card, return result; } +#if IS_ENABLED(CONFIG_SND_CTL_DEBUG) + +static const char *const snd_ctl_elem_iface_names[] = { + [SNDRV_CTL_ELEM_IFACE_CARD] = "CARD", + [SNDRV_CTL_ELEM_IFACE_HWDEP] = "HWDEP", + [SNDRV_CTL_ELEM_IFACE_MIXER] = "MIXER", + [SNDRV_CTL_ELEM_IFACE_PCM] = "PCM", + [SNDRV_CTL_ELEM_IFACE_RAWMIDI] = "RAWMIDI", + [SNDRV_CTL_ELEM_IFACE_TIMER] = "TIMER", + [SNDRV_CTL_ELEM_IFACE_SEQUENCER] = "SEQUENCER", +}; + +static int snd_ctl_put_verify(struct snd_card *card, struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *control) +{ + struct snd_ctl_elem_value *original = card->value_buf; + struct snd_ctl_elem_info info; + const char *iname; + int ret, retcmp; + + memset(original, 0, sizeof(*original)); + memset(&info, 0, sizeof(info)); + + ret = kctl->info(kctl, &info); + if (ret) + return ret; + + ret = kctl->get(kctl, original); + if (ret) + return ret; + + ret = kctl->put(kctl, control); + if (ret < 0) + return ret; + + /* Sanitize the new value (control->value) before comparing. */ + fill_remaining_elem_value(control, &info, 0); + + /* With known state for both new and original, do the comparison. */ + retcmp = memcmp(&original->value, &control->value, sizeof(original->value)); + if (retcmp) + retcmp = 1; + + iname = snd_ctl_elem_iface_names[kctl->id.iface]; + trace_snd_ctl_put(&kctl->id, iname, card->number, ret, retcmp); + + return ret; +} + +static int snd_ctl_put(struct snd_card *card, struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *control, unsigned int access) +{ + if ((access & SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK) || + (access & SNDRV_CTL_ELEM_ACCESS_VOLATILE)) + return kctl->put(kctl, control); + + return snd_ctl_put_verify(card, kctl, control); +} +#else +static inline int snd_ctl_put(struct snd_card *card, struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *control, unsigned int access) +{ + return kctl->put(kctl, control); +} +#endif + static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_ctl_elem_value *control) { @@ -1300,7 +1373,8 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, false); } if (!result) - result = kctl->put(kctl, control); + result = snd_ctl_put(card, kctl, control, vd->access); + if (result < 0) { up_write(&card->controls_rwsem); return result; diff --git a/sound/core/control_trace.h b/sound/core/control_trace.h new file mode 100644 index 000000000000..d30e654b0860 --- /dev/null +++ b/sound/core/control_trace.h @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_ctl + +#if !defined(_TRACE_SND_CTL_H) || defined(TRACE_HEADER_MULTI_READ) +#define _TRACE_SND_CTL_H + +#include <linux/tracepoint.h> +#include <uapi/sound/asound.h> + +TRACE_EVENT(snd_ctl_put, + + TP_PROTO(struct snd_ctl_elem_id *id, const char *iname, unsigned int card, + int expected, int actual), + + TP_ARGS(id, iname, card, expected, actual), + + TP_STRUCT__entry( + __field(unsigned int, numid) + __string(iname, iname) + __string(kname, id->name) + __field(unsigned int, index) + __field(unsigned int, device) + __field(unsigned int, subdevice) + __field(unsigned int, card) + __field(int, expected) + __field(int, actual) + ), + + TP_fast_assign( + __entry->numid = id->numid; + __assign_str(iname); + __assign_str(kname); + __entry->index = id->index; + __entry->device = id->device; + __entry->subdevice = id->subdevice; + __entry->card = card; + __entry->expected = expected; + __entry->actual = actual; + ), + + TP_printk("%s: expected=%d, actual=%d for ctl numid=%d, iface=%s, name='%s', index=%d, device=%d, subdevice=%d, card=%d\n", + __entry->expected == __entry->actual ? "success" : "fail", + __entry->expected, __entry->actual, __entry->numid, + __get_str(iname), __get_str(kname), __entry->index, + __entry->device, __entry->subdevice, __entry->card) +); + +#endif /* _TRACE_SND_CTL_H */ + +/* This part must be outside protection */ +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#define TRACE_INCLUDE_FILE control_trace +#include <trace/define_trace.h> diff --git a/sound/core/init.c b/sound/core/init.c index 2f1bd9cbdbed..0c316189e947 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -363,6 +363,11 @@ static int snd_card_init(struct snd_card *card, struct device *parent, card->debugfs_root = debugfs_create_dir(dev_name(&card->card_dev), sound_debugfs_root); #endif +#ifdef CONFIG_SND_CTL_DEBUG + card->value_buf = kmalloc(sizeof(*card->value_buf), GFP_KERNEL); + if (!card->value_buf) + return -ENOMEM; +#endif return 0; __error_ctl: @@ -587,6 +592,9 @@ static int snd_card_do_free(struct snd_card *card) snd_device_free_all(card); if (card->private_free) card->private_free(card); +#ifdef CONFIG_SND_CTL_DEBUG + kfree(card->value_buf); +#endif if (snd_info_card_free(card) < 0) { dev_warn(card->dev, "unable to free card info\n"); /* Not fatal error */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d4fd4dfc7fc3..a140a0d9abb8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1227,14 +1227,16 @@ static int snd_pcm_oss_capture_position_fixup(struct snd_pcm_substream *substrea snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const char *ptr, snd_pcm_uframes_t frames, int in_kernel) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_state_t state; int ret; while (1) { - if (runtime->state == SNDRV_PCM_STATE_XRUN || - runtime->state == SNDRV_PCM_STATE_SUSPENDED) { + state = snd_pcm_get_state(substream); + if (state == SNDRV_PCM_STATE_XRUN || + state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: write: recovering from %s\n", - runtime->state == SNDRV_PCM_STATE_XRUN ? + state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_oss_prepare(substream); @@ -1249,7 +1251,7 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const break; /* test, if we can't store new data, because the stream */ /* has not been started */ - if (runtime->state == SNDRV_PCM_STATE_PREPARED) + if (snd_pcm_get_state(substream) == SNDRV_PCM_STATE_PREPARED) return -EAGAIN; } return ret; @@ -1259,20 +1261,22 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t delay; + snd_pcm_state_t state; int ret; while (1) { - if (runtime->state == SNDRV_PCM_STATE_XRUN || - runtime->state == SNDRV_PCM_STATE_SUSPENDED) { + state = snd_pcm_get_state(substream); + if (state == SNDRV_PCM_STATE_XRUN || + state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: read: recovering from %s\n", - runtime->state == SNDRV_PCM_STATE_XRUN ? + state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) break; - } else if (runtime->state == SNDRV_PCM_STATE_SETUP) { + } else if (state == SNDRV_PCM_STATE_SETUP) { ret = snd_pcm_oss_prepare(substream); if (ret < 0) break; @@ -1285,7 +1289,7 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p frames, in_kernel); mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { - if (runtime->state == SNDRV_PCM_STATE_DRAINING) { + if (snd_pcm_get_state(substream) == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); if (ret < 0) break; @@ -1301,15 +1305,16 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p #ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames) { - struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_state_t state; int ret; while (1) { - if (runtime->state == SNDRV_PCM_STATE_XRUN || - runtime->state == SNDRV_PCM_STATE_SUSPENDED) { + state = snd_pcm_get_state(substream); + if (state == SNDRV_PCM_STATE_XRUN || + state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: writev: recovering from %s\n", - runtime->state == SNDRV_PCM_STATE_XRUN ? + state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_oss_prepare(substream); @@ -1322,7 +1327,7 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void /* test, if we can't store new data, because the stream */ /* has not been started */ - if (runtime->state == SNDRV_PCM_STATE_PREPARED) + if (snd_pcm_get_state(substream) == SNDRV_PCM_STATE_PREPARED) return -EAGAIN; } return ret; @@ -1330,21 +1335,22 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames) { - struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_state_t state; int ret; while (1) { - if (runtime->state == SNDRV_PCM_STATE_XRUN || - runtime->state == SNDRV_PCM_STATE_SUSPENDED) { + state = snd_pcm_get_state(substream); + if (state == SNDRV_PCM_STATE_XRUN || + state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "pcm_oss: readv: recovering from %s\n", - runtime->state == SNDRV_PCM_STATE_XRUN ? + state == SNDRV_PCM_STATE_XRUN ? "XRUN" : "SUSPEND"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) break; - } else if (runtime->state == SNDRV_PCM_STATE_SETUP) { + } else if (state == SNDRV_PCM_STATE_SETUP) { ret = snd_pcm_oss_prepare(substream); if (ret < 0) break; diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index e71f393d3b01..5313f50f17da 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -430,11 +430,13 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream, if (!boundary) boundary = 0x7fffffff; scoped_guard(pcm_stream_lock_irq, substream) { - /* FIXME: we should consider the boundary for the sync from app */ - if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) - control->appl_ptr = scontrol.appl_ptr; - else + if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, scontrol.appl_ptr); + if (err < 0) + return err; + } else { scontrol.appl_ptr = control->appl_ptr % boundary; + } if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) control->avail_min = scontrol.avail_min; else diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5a64453da728..a541bb235cfa 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -618,13 +618,32 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } -static void snd_pcm_set_state(struct snd_pcm_substream *substream, - snd_pcm_state_t state) +/** + * snd_pcm_set_state - Set the PCM runtime state with stream lock + * @substream: PCM substream + * @state: state to set + */ +void snd_pcm_set_state(struct snd_pcm_substream *substream, + snd_pcm_state_t state) { guard(pcm_stream_lock_irq)(substream); if (substream->runtime->state != SNDRV_PCM_STATE_DISCONNECTED) __snd_pcm_set_state(substream->runtime, state); } +EXPORT_SYMBOL_GPL(snd_pcm_set_state); + +/** + * snd_pcm_get_state - Read the PCM runtime state with stream lock + * @substream: PCM substream + * + * Return: the current PCM state + */ +snd_pcm_state_t snd_pcm_get_state(struct snd_pcm_substream *substream) +{ + guard(pcm_stream_lock_irqsave)(substream); + return substream->runtime->state; +} +EXPORT_SYMBOL_GPL(snd_pcm_get_state); static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, int event) @@ -1761,6 +1780,9 @@ static int snd_pcm_suspend(struct snd_pcm_substream *substream) * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * + * Takes and releases pcm->open_mutex to serialize against + * concurrent open/close while walking the substreams. + * * After this call, all streams are changed to SUSPENDED state. * * Return: Zero if successful (or @pcm is %NULL), or a negative error code. @@ -1773,8 +1795,9 @@ int snd_pcm_suspend_all(struct snd_pcm *pcm) if (! pcm) return 0; + guard(mutex)(&pcm->open_mutex); + for_each_pcm_substream(pcm, stream, substream) { - /* FIXME: the open/close code should lock this as well */ if (!substream->runtime) continue; diff --git a/sound/core/seq/oss/seq_oss_rw.c b/sound/core/seq/oss/seq_oss_rw.c index 8a142fd54a19..307ef98c44c7 100644 --- a/sound/core/seq/oss/seq_oss_rw.c +++ b/sound/core/seq/oss/seq_oss_rw.c @@ -101,9 +101,9 @@ snd_seq_oss_write(struct seq_oss_devinfo *dp, const char __user *buf, int count, break; } fmt = (*(unsigned short *)rec.c) & 0xffff; - /* FIXME the return value isn't correct */ - return snd_seq_oss_synth_load_patch(dp, rec.s.dev, - fmt, buf, 0, count); + err = snd_seq_oss_synth_load_patch(dp, rec.s.dev, + fmt, buf, 0, count); + return err < 0 ? err : count; } if (ev_is_long(&rec)) { /* extended code */ diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index db2f169cae11..ff4ee26adad1 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -841,7 +841,7 @@ static int cc_ev_to_ump_midi2(const struct snd_seq_event *event, unsigned char index = event->data.control.param & 0x7f; unsigned char val = event->data.control.value & 0x7f; struct ump_cvt_to_ump_bank *cc = &dest_port->midi2_bank[channel]; - int ret; + int ret = 0; /* process special CC's (bank/rpn/nrpn) */ switch (index) { @@ -851,47 +851,54 @@ static int cc_ev_to_ump_midi2(const struct snd_seq_event *event, cc->cc_rpn_msb = val; if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f) reset_rpn(cc); - return ret; + break; case UMP_CC_RPN_LSB: ret = fill_rpn(cc, data, channel, true); cc->rpn_set = 1; cc->cc_rpn_lsb = val; if (cc->cc_rpn_msb == 0x7f && cc->cc_rpn_lsb == 0x7f) reset_rpn(cc); - return ret; + break; case UMP_CC_NRPN_MSB: ret = fill_rpn(cc, data, channel, true); cc->nrpn_set = 1; cc->cc_nrpn_msb = val; - return ret; + break; case UMP_CC_NRPN_LSB: ret = fill_rpn(cc, data, channel, true); cc->nrpn_set = 1; cc->cc_nrpn_lsb = val; - return ret; + break; case UMP_CC_DATA: cc->cc_data_msb_set = 1; cc->cc_data_msb = val; - return fill_rpn(cc, data, channel, false); + ret = fill_rpn(cc, data, channel, false); + break; case UMP_CC_BANK_SELECT: cc->bank_set = 1; cc->cc_bank_msb = val; - return 0; // skip + ret = 0; // skip + break; case UMP_CC_BANK_SELECT_LSB: cc->bank_set = 1; cc->cc_bank_lsb = val; - return 0; // skip + ret = 0; // skip + break; case UMP_CC_DATA_LSB: cc->cc_data_lsb_set = 1; cc->cc_data_lsb = val; - return fill_rpn(cc, data, channel, false); + ret = fill_rpn(cc, data, channel, false); + break; + default: + data->cc.status = status; + data->cc.channel = channel; + data->cc.index = index; + data->cc.data = upscale_7_to_32bit(event->data.control.value & 0x7f); + ret = 1; + break; } - data->cc.status = status; - data->cc.channel = channel; - data->cc.index = index; - data->cc.data = upscale_7_to_32bit(event->data.control.value & 0x7f); - return 1; + return ret; } /* convert one-parameter control event to MIDI 2.0 UMP */ diff --git a/sound/core/sound.c b/sound/core/sound.c index 93436db24710..8d05fe0d263b 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -216,9 +216,16 @@ static int snd_find_free_minor(int type, struct snd_card *card, int dev) case SNDRV_DEVICE_TYPE_RAWMIDI: case SNDRV_DEVICE_TYPE_PCM_PLAYBACK: case SNDRV_DEVICE_TYPE_PCM_CAPTURE: + if (snd_BUG_ON(!card)) + return -EINVAL; + minor = SNDRV_MINOR(card->number, type + dev); + break; case SNDRV_DEVICE_TYPE_COMPRESS: if (snd_BUG_ON(!card)) return -EINVAL; + if (dev < 0 || + dev >= SNDRV_MINOR_HWDEP - SNDRV_MINOR_COMPRESS) + return -EINVAL; minor = SNDRV_MINOR(card->number, type + dev); break; default: diff --git a/sound/core/timer.c b/sound/core/timer.c index 6a70df7ae019..820901d503af 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -129,6 +129,9 @@ static LIST_HEAD(snd_timer_list); /* list of slave instances */ static LIST_HEAD(snd_timer_slave_list); +/* list of open master instances that can accept slave links */ +static LIST_HEAD(snd_timer_master_list); + /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); @@ -161,6 +164,7 @@ struct snd_timer_instance *snd_timer_instance_new(const char *owner) } INIT_LIST_HEAD(&timeri->open_list); INIT_LIST_HEAD(&timeri->active_list); + INIT_LIST_HEAD(&timeri->master_list); INIT_LIST_HEAD(&timeri->ack_list); INIT_LIST_HEAD(&timeri->slave_list_head); INIT_LIST_HEAD(&timeri->slave_active_head); @@ -245,6 +249,12 @@ static int check_matching_master_slave(struct snd_timer_instance *master, return 1; } +static bool snd_timer_has_slave_key(const struct snd_timer_instance *timeri) +{ + return !(timeri->flags & SNDRV_TIMER_IFLG_SLAVE) && + timeri->slave_class > SNDRV_TIMER_SCLASS_NONE; +} + /* * look for a master instance matching with the slave id of the given slave. * when found, relink the open_link of the slave. @@ -253,19 +263,15 @@ static int check_matching_master_slave(struct snd_timer_instance *master, */ static int snd_timer_check_slave(struct snd_timer_instance *slave) { - struct snd_timer *timer; struct snd_timer_instance *master; int err = 0; - /* FIXME: it's really dumb to look up all entries.. */ - list_for_each_entry(timer, &snd_timer_list, device_list) { - list_for_each_entry(master, &timer->open_list_head, open_list) { - err = check_matching_master_slave(master, slave); - if (err != 0) /* match found or error */ - goto out; - } + list_for_each_entry(master, &snd_timer_master_list, master_list) { + err = check_matching_master_slave(master, slave); + if (err != 0) /* match found or error */ + goto out; } - out: +out: return err < 0 ? err : 0; } @@ -377,6 +383,8 @@ int snd_timer_open(struct snd_timer_instance *timeri, timeri->slave_id = slave_id; list_add_tail(&timeri->open_list, &timer->open_list_head); + if (snd_timer_has_slave_key(timeri)) + list_add_tail(&timeri->master_list, &snd_timer_master_list); timer->num_instances++; err = snd_timer_check_master(timeri); list_added: @@ -431,6 +439,9 @@ static void snd_timer_close_locked(struct snd_timer_instance *timeri, num_slaves--; } + if (!list_empty(&timeri->master_list)) + list_del_init(&timeri->master_list); + /* force to stop the timer */ snd_timer_stop(timeri); diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c index 2b595ee0bc35..05550f36fac5 100644 --- a/sound/firewire/fireworks/fireworks_command.c +++ b/sound/firewire/fireworks/fireworks_command.c @@ -151,10 +151,13 @@ efw_transaction(struct snd_efw *efw, unsigned int category, (be32_to_cpu(header->category) != category) || (be32_to_cpu(header->command) != command) || (be32_to_cpu(header->status) != EFR_STATUS_OK)) { + u32 st = be32_to_cpu(header->status); + dev_err(&efw->unit->device, "EFW command failed [%u/%u]: %s\n", be32_to_cpu(header->category), be32_to_cpu(header->command), - efr_status_names[be32_to_cpu(header->status)]); + st < ARRAY_SIZE(efr_status_names) ? + efr_status_names[st] : "unknown"); err = -EIO; goto end; } diff --git a/sound/hda/codecs/analog.c b/sound/hda/codecs/analog.c index 11b1d30b23fd..1ba8ae54e25e 100644 --- a/sound/hda/codecs/analog.c +++ b/sound/hda/codecs/analog.c @@ -38,6 +38,8 @@ struct ad198x_spec { unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ int num_smux_conns; + + unsigned int gpio_data; }; @@ -934,9 +936,9 @@ static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) if (spec->eapd_nid) ad_vmaster_eapd_hook(private_data, enabled); - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, - enabled ? 0x00 : 0x02); + spec->gpio_data = enabled ? 0x00 : 0x02; + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, spec->gpio_data); } static void ad1884_fixup_hp_eapd(struct hda_codec *codec, @@ -948,12 +950,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; spec->gen.own_eapd_ctl = 1; - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_MASK, 0x02); - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x02); - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 0x02); + spec->gpio_data = 0x02; break; case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) @@ -961,6 +958,9 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, else spec->eapd_nid = spec->gen.autocfg.speaker_pins[0]; break; + case HDA_FIXUP_ACT_INIT: + snd_hda_codec_set_gpio(codec, 0x02, 0x02, spec->gpio_data, 0); + break; } } diff --git a/sound/hda/codecs/ca0132.c b/sound/hda/codecs/ca0132.c index a0677d7da8e2..ad533b04ab29 100644 --- a/sound/hda/codecs/ca0132.c +++ b/sound/hda/codecs/ca0132.c @@ -3755,22 +3755,12 @@ static void ca0132_gpio_setup(struct hda_codec *codec) switch (ca0132_quirk(spec)) { case QUIRK_SBZ: - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x07); - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_MASK, 0x07); - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 0x04); + snd_hda_codec_set_gpio(codec, 0x07, 0x07, 0x04, 0); snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x06); break; case QUIRK_R3DI: - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x1E); - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_MASK, 0x1F); - snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 0x0C); + snd_hda_codec_set_gpio(codec, 0x1F, 0x1E, 0x0C, 0); break; default: break; diff --git a/sound/hda/codecs/cirrus/cs420x.c b/sound/hda/codecs/cirrus/cs420x.c index 52eda0a338ea..42559edbba05 100644 --- a/sound/hda/codecs/cirrus/cs420x.c +++ b/sound/hda/codecs/cirrus/cs420x.c @@ -264,14 +264,9 @@ static int cs_init(struct hda_codec *codec) snd_hda_gen_init(codec); - if (spec->gpio_mask) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, - spec->gpio_mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, - spec->gpio_dir); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_data); - } + if (spec->gpio_mask) + snd_hda_codec_set_gpio(codec, spec->gpio_mask, spec->gpio_dir, + spec->gpio_data, 0); if (spec->vendor_nid == CS420X_VENDOR_NID) { init_input_coef(codec); diff --git a/sound/hda/codecs/cirrus/cs421x.c b/sound/hda/codecs/cirrus/cs421x.c index c8349a2c5a36..645b06599e5f 100644 --- a/sound/hda/codecs/cirrus/cs421x.c +++ b/sound/hda/codecs/cirrus/cs421x.c @@ -442,14 +442,9 @@ static int cs421x_init(struct hda_codec *codec) snd_hda_gen_init(codec); - if (spec->gpio_mask) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, - spec->gpio_mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, - spec->gpio_dir); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_data); - } + if (spec->gpio_mask) + snd_hda_codec_set_gpio(codec, spec->gpio_mask, spec->gpio_dir, + spec->gpio_data, 0); cs4210_spdif_automute(codec, NULL); diff --git a/sound/hda/codecs/cirrus/cs8409.c b/sound/hda/codecs/cirrus/cs8409.c index fad705092777..c43ff3ef75b6 100644 --- a/sound/hda/codecs/cirrus/cs8409.c +++ b/sound/hda/codecs/cirrus/cs8409.c @@ -268,7 +268,7 @@ static int cs8409_i2c_bulk_read(struct sub_codec *scodec, struct cs8409_i2c_para return 0; error: - codec_err(codec, "I2C Bulk Write Failed 0x%02x\n", scodec->addr); + codec_err(codec, "I2C Bulk Read Failed 0x%02x\n", scodec->addr); return -EIO; } @@ -1042,14 +1042,9 @@ static void cs8409_cs42l42_hw_init(struct hda_codec *codec) struct cs8409_spec *spec = codec->spec; struct sub_codec *cs42l42 = spec->scodecs[CS8409_CODEC0]; - if (spec->gpio_mask) { - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_MASK, - spec->gpio_mask); - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DIRECTION, - spec->gpio_dir); - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_data); - } + if (spec->gpio_mask) + snd_hda_codec_set_gpio(codec, spec->gpio_mask, spec->gpio_dir, + spec->gpio_data, 0); for (; seq->nid; seq++) cs8409_vendor_coef_set(codec, seq->cir, seq->coeff); @@ -1442,14 +1437,9 @@ static void dolphin_hw_init(struct hda_codec *codec) struct sub_codec *cs42l42; int i; - if (spec->gpio_mask) { - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_MASK, - spec->gpio_mask); - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DIRECTION, - spec->gpio_dir); - snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_data); - } + if (spec->gpio_mask) + snd_hda_codec_set_gpio(codec, spec->gpio_mask, spec->gpio_dir, + spec->gpio_data, 0); for (; seq->nid; seq++) cs8409_vendor_coef_set(codec, seq->cir, seq->coeff); diff --git a/sound/hda/codecs/cmedia.c b/sound/hda/codecs/cmedia.c index e6e12c01339f..88dd80d987d4 100644 --- a/sound/hda/codecs/cmedia.c +++ b/sound/hda/codecs/cmedia.c @@ -46,13 +46,6 @@ static int cmedia_probe(struct hda_codec *codec, const struct hda_device_id *id) if (err < 0) goto error; - err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); - if (err < 0) - goto error; - err = snd_hda_gen_parse_auto_config(codec, cfg); - if (err < 0) - goto error; - if (is_cmi8888) { if (get_defcfg_device(snd_hda_codec_get_pincfg(codec, 0x10)) == AC_JACK_HP_OUT) { diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index aa726eb323eb..3a9717df39b4 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -154,14 +154,8 @@ static void cxt_init_gpio_led(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; - if (mask) { - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, - mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, - mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); - } + if (mask) + snd_hda_codec_set_gpio(codec, mask, mask, spec->gpio_led, 0); } static void cx_fixup_headset_recog(struct hda_codec *codec) @@ -775,9 +769,7 @@ static void cxt_setup_gpio_unmute(struct hda_codec *codec, { if (gpio_mute_mask) { // set gpio data to 0. - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, gpio_mute_mask); - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, gpio_mute_mask); + snd_hda_codec_set_gpio(codec, gpio_mute_mask, gpio_mute_mask, 0, 0); snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_STICKY_MASK, 0); } } diff --git a/sound/hda/codecs/generic.c b/sound/hda/codecs/generic.c index 092428ada29d..660a9f2c0ded 100644 --- a/sound/hda/codecs/generic.c +++ b/sound/hda/codecs/generic.c @@ -863,7 +863,7 @@ static void sync_power_state_change(struct hda_codec *codec, hda_nid_t nid) { if (nid) { msleep(10); - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + snd_hda_codec_write_sync(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); } } diff --git a/sound/hda/codecs/hdmi/intelhdmi.c b/sound/hda/codecs/hdmi/intelhdmi.c index 9460c8db39a9..6a7882544ab7 100644 --- a/sound/hda/codecs/hdmi/intelhdmi.c +++ b/sound/hda/codecs/hdmi/intelhdmi.c @@ -85,7 +85,7 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, } } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); + snd_hda_codec_write_sync(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state); } diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 45f9d6487388..0a5eadec3ef6 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -1004,13 +1004,6 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_regmap_sync(codec); hda_call_check_power_status(codec, 0x01); - /* on some machine, the BIOS will clear the codec gpio data when enter - * suspend, and won't restore the data after resume, so we restore it - * in the driver. - */ - if (spec->gpio_data) - alc_write_gpio_data(codec); - if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -2296,9 +2289,9 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } - else + } else { alc_fixup_headset_mode(codec, fix, action); + } } static void alc288_update_headset_jack_cb(struct hda_codec *codec, @@ -6639,10 +6632,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc288_fixup_surface_swap_dacs, }, - [ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY] = { - .type = HDA_FIXUP_FUNC, - .v.func = alc233_fixup_lenovo_gpio2_mic_hotkey, - }, + [ALC233_FIXUP_LENOVO_GPIO2_MIC_HOTKEY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_lenovo_gpio2_mic_hotkey, + }, [ALC245_FIXUP_BASS_HP_DAC] = { .type = HDA_FIXUP_FUNC, /* Borrow the DAC routing selected for those Thinkpads */ @@ -7198,6 +7191,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e60, "HP OmniBook 7 Laptop 16-bh0xxx", ALC245_FIXUP_CS35L41_I2C_2_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8e61, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e62, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8e75, "HP Trekker G7JC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e8a, "HP NexusX", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8e9c, "HP 16 Clipper OmniBook X X360", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e9d, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), @@ -7219,8 +7213,11 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8ee4, "HP Bantie A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8ee5, "HP Bantie A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), SND_PCI_QUIRK(0x103c, 0x8ee7, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), + SND_PCI_QUIRK(0x103c, 0x8f07, "HP Agusta G7KX", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8f0c, "HP ZBook X G2i 16W", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8f0e, "HP ZBook X G2i 16W", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8f2d, "HP Auster 14", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8f2e, "HP Auster 14", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8f40, "HP ZBook 8 G2a 14", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8f41, "HP ZBook 8 G2a 16", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8f42, "HP ZBook 8 G2a 14W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), @@ -7365,6 +7362,10 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x31e1, "ASUS B5605CCA", ALC294_FIXUP_ASUS_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x31f1, "ASUS B3605CCA", ALC294_FIXUP_ASUS_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3391, "ASUS PM3606CKA", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x3601, "ASUS PM5406CGA", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x3611, "ASUS PM5606CGA", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x1043, 0x3701, "ASUS P5406CCA", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3711, "ASUS P5606CCA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), diff --git a/sound/hda/codecs/realtek/alc662.c b/sound/hda/codecs/realtek/alc662.c index 5073165d1f3c..2cf3664a35ff 100644 --- a/sound/hda/codecs/realtek/alc662.c +++ b/sound/hda/codecs/realtek/alc662.c @@ -255,6 +255,17 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +static void alc662_fixup_csl_amp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_INIT) { + /* need to toggle GPIO to enable the amp */ + snd_hda_codec_set_gpio(codec, 0x03, 0x03, 0x03, 0); + msleep(100); + snd_hda_codec_set_gpio(codec, 0x03, 0x03, 0x00, 0); + } +} + enum { ALC662_FIXUP_ASPIRE, ALC662_FIXUP_LED_GPIO1, @@ -313,6 +324,7 @@ enum { ALC897_FIXUP_HEADSET_MIC_PIN2, ALC897_FIXUP_UNIS_H3C_X500S, ALC897_FIXUP_HEADSET_MIC_PIN3, + ALC662_FIXUP_CSL_GPIO, }; static const struct hda_fixup alc662_fixups[] = { @@ -766,11 +778,16 @@ static const struct hda_fixup alc662_fixups[] = { { } }, }, + [ALC662_FIXUP_CSL_GPIO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_csl_amp, + }, }; static const struct hda_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1019, 0x9859, "JP-IK LEAP W502", ALC897_FIXUP_HEADSET_MIC_PIN3), + SND_PCI_QUIRK(0x1022, 0xc950, "CSL Unity BF24B", ALC662_FIXUP_CSL_GPIO), SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), diff --git a/sound/hda/codecs/realtek/realtek.c b/sound/hda/codecs/realtek/realtek.c index aad265c0a5b6..db365a746b1a 100644 --- a/sound/hda/codecs/realtek/realtek.c +++ b/sound/hda/codecs/realtek/realtek.c @@ -99,15 +99,6 @@ void alc_setup_gpio(struct hda_codec *codec, unsigned int mask) } EXPORT_SYMBOL_NS_GPL(alc_setup_gpio, "SND_HDA_CODEC_REALTEK"); -void alc_write_gpio_data(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_data); -} -EXPORT_SYMBOL_NS_GPL(alc_write_gpio_data, "SND_HDA_CODEC_REALTEK"); - void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask, bool on) { @@ -119,26 +110,22 @@ void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask, else spec->gpio_data &= ~mask; if (oldval != spec->gpio_data) - alc_write_gpio_data(codec); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); } EXPORT_SYMBOL_NS_GPL(alc_update_gpio_data, "SND_HDA_CODEC_REALTEK"); -void alc_write_gpio(struct hda_codec *codec) +static void alc_write_gpio(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; if (!spec->gpio_mask) return; - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_MASK, spec->gpio_mask); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DIRECTION, spec->gpio_dir); - if (spec->gpio_write_delay) - msleep(1); - alc_write_gpio_data(codec); + snd_hda_codec_set_gpio(codec, spec->gpio_mask, spec->gpio_dir, + spec->gpio_data, + spec->gpio_write_delay ? 1 : 0); } -EXPORT_SYMBOL_NS_GPL(alc_write_gpio, "SND_HDA_CODEC_REALTEK"); void alc_fixup_gpio(struct hda_codec *codec, int action, unsigned int mask) { @@ -411,9 +398,8 @@ void alc_headset_mic_no_shutup(struct hda_codec *codec) return; snd_array_for_each(&codec->init_pins, i, pin) { - /* use read here for syncing after issuing each verb */ if (pin->nid != mic_pin) - snd_hda_codec_read(codec, pin->nid, 0, + snd_hda_codec_write_sync(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); } @@ -2164,8 +2150,7 @@ void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; - } - else + } else alc_fixup_headset_mode(codec, fix, action); } EXPORT_SYMBOL_NS_GPL(alc_fixup_headset_mode_no_hp_mic, "SND_HDA_CODEC_REALTEK"); diff --git a/sound/hda/codecs/realtek/realtek.h b/sound/hda/codecs/realtek/realtek.h index b2a919904c4c..de95642bb648 100644 --- a/sound/hda/codecs/realtek/realtek.h +++ b/sound/hda/codecs/realtek/realtek.h @@ -168,10 +168,8 @@ void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw); * GPIO helpers */ void alc_setup_gpio(struct hda_codec *codec, unsigned int mask); -void alc_write_gpio_data(struct hda_codec *codec); void alc_update_gpio_data(struct hda_codec *codec, unsigned int mask, bool on); -void alc_write_gpio(struct hda_codec *codec); /* common GPIO fixups */ void alc_fixup_gpio(struct hda_codec *codec, int action, unsigned int mask); diff --git a/sound/hda/codecs/senarytech.c b/sound/hda/codecs/senarytech.c index 6239a25bb8f3..3ee8bc0ea3ab 100644 --- a/sound/hda/codecs/senarytech.c +++ b/sound/hda/codecs/senarytech.c @@ -36,6 +36,32 @@ struct senary_spec { unsigned int gpio_mic_led_mask; }; +enum { + SENARY_FIXUP_PINCFG_DEFAULT, +}; + +static const struct hda_pintbl senary_pincfg_default[] = { + { 0x16, 0x02211020 }, /* Headphone */ + { 0x17, 0x40f001f0 }, /* Not used */ + { 0x18, 0x05a1904d }, /* Mic */ + { 0x19, 0x02a1104e }, /* Headset Mic */ + { 0x1a, 0x01819030 }, /* Line-in */ + { 0x1d, 0x01014010 }, /* Line-out */ + {} +}; + +static const struct hda_fixup senary_fixups[] = { + [SENARY_FIXUP_PINCFG_DEFAULT] = { + .type = HDA_FIXUP_PINS, + .v.pins = senary_pincfg_default, + }, +}; + +/* Quirk table for specific machines can be added here */ +static const struct hda_quirk sn6186_fixups[] = { + {} +}; + #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; private_value will be overwritten */ static const struct snd_kcontrol_new senary_beep_mixer[] = { @@ -50,7 +76,6 @@ static int set_beep_amp(struct senary_spec *spec, hda_nid_t nid, unsigned int beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir); int i; - spec->gen.beep_nid = nid; for (i = 0; i < ARRAY_SIZE(senary_beep_mixer); i++) { knew = snd_hda_gen_add_kctl(&spec->gen, NULL, &senary_beep_mixer[i]); @@ -58,6 +83,8 @@ static int set_beep_amp(struct senary_spec *spec, hda_nid_t nid, return -ENOMEM; knew->private_value = beep_amp; } + + spec->gen.beep_nid = nid; return 0; } @@ -93,16 +120,28 @@ static void senary_auto_parse_eapd(struct hda_codec *codec) } } +/* Hardware specific initialization verbs */ +static void senary_init_verb(struct hda_codec *codec) +{ + /* Vendor specific init sequence */ + snd_hda_codec_write(codec, 0x1b, 0x0, 0x05a, 0xaa); + snd_hda_codec_write(codec, 0x1b, 0x0, 0x059, 0x48); + snd_hda_codec_write(codec, 0x1b, 0x0, 0x01b, 0x00); + snd_hda_codec_write(codec, 0x1b, 0x0, 0x01c, 0x00); + + /* Override pin caps for headset mic */ + snd_hda_override_pin_caps(codec, 0x19, 0x2124); +} + static void senary_auto_turn_eapd(struct hda_codec *codec, int num_pins, const hda_nid_t *pins, bool on) { int i; for (i = 0; i < num_pins; i++) { - if (snd_hda_query_pin_caps(codec, pins[i]) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_EAPD_BTLENABLE, - on ? 0x02 : 0); + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_EAPD_BTLENABLE, + on ? 0x02 : 0); } } @@ -120,14 +159,8 @@ static void senary_init_gpio_led(struct hda_codec *codec) struct senary_spec *spec = codec->spec; unsigned int mask = spec->gpio_mute_led_mask | spec->gpio_mic_led_mask; - if (mask) { - snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_MASK, - mask); - snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DIRECTION, - mask); - snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_GPIO_DATA, - spec->gpio_led); - } + if (mask) + snd_hda_codec_set_gpio(codec, mask, mask, spec->gpio_led, 0); } static int senary_init(struct hda_codec *codec) @@ -136,6 +169,7 @@ static int senary_init(struct hda_codec *codec) snd_hda_gen_init(codec); senary_init_gpio_led(codec); + senary_init_verb(codec); if (!spec->dynamic_eapd) senary_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); @@ -181,11 +215,30 @@ static int senary_probe(struct hda_codec *codec, const struct hda_device_id *id) senary_auto_parse_eapd(codec); spec->gen.own_eapd_ctl = 1; - if (!spec->gen.vmaster_mute.hook) - spec->gen.vmaster_mute.hook = senary_auto_vmaster_hook; + /* Setup fixups based on codec vendor ID */ + switch (codec->core.vendor_id) { + case 0x1fa86186: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x15; + snd_hda_pick_fixup(codec, NULL, sn6186_fixups, senary_fixups); + + /* If no specific quirk found, apply the default pin configuration */ + if (codec->fixup_id == HDA_FIXUP_ID_NOT_SET) + codec->fixup_id = SENARY_FIXUP_PINCFG_DEFAULT; + break; + default: + snd_hda_pick_fixup(codec, NULL, sn6186_fixups, senary_fixups); + break; + } snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + /* Run hardware init verbs once during probe */ + senary_init_verb(codec); + + if (!spec->gen.vmaster_mute.hook) + spec->gen.vmaster_mute.hook = senary_auto_vmaster_hook; + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, spec->parse_flags); if (err < 0) diff --git a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c index 16d5ea77192f..732ae534db36 100644 --- a/sound/hda/codecs/side-codecs/cs35l41_hda_property.c +++ b/sound/hda/codecs/side-codecs/cs35l41_hda_property.c @@ -55,6 +55,11 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "103C8A30", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8A31", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8A6E", 4, EXTERNAL, { CS35L41_LEFT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_RIGHT }, 0, -1, -1, 0, 0, 0 }, +/* + * Device 103C8B63 has _DSD with valid reset-gpios and cs-gpios, however the + * boost type is incorrectly set to Internal. Override to External Boost. + */ + { "103C8B63", 4, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT }, -1, -1, -1, 0, 0, 0 }, { "103C8BB3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BB4", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BDD", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, @@ -475,6 +480,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "103C8A30", generic_dsd_config }, { "CSC3551", "103C8A31", generic_dsd_config }, { "CSC3551", "103C8A6E", generic_dsd_config }, + { "CSC3551", "103C8B63", generic_dsd_config }, { "CSC3551", "103C8BB3", generic_dsd_config }, { "CSC3551", "103C8BB4", generic_dsd_config }, { "CSC3551", "103C8BDD", generic_dsd_config }, diff --git a/sound/hda/codecs/sigmatel.c b/sound/hda/codecs/sigmatel.c index acbbc7c3508b..ee3bd21adc36 100644 --- a/sound/hda/codecs/sigmatel.c +++ b/sound/hda/codecs/sigmatel.c @@ -309,15 +309,7 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, fg, 0, 0x7e7, 0); - snd_hda_codec_write(codec, fg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_read(codec, fg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ - - msleep(1); - - snd_hda_codec_read(codec, fg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ + snd_hda_codec_set_gpio(codec, gpiomask, gpiodir, gpiostate, 1); } /* hook for controlling mic-mute LED GPIO */ diff --git a/sound/hda/common/codec.c b/sound/hda/common/codec.c index 09b1329bb8f3..c2af2511a831 100644 --- a/sound/hda/common/codec.c +++ b/sound/hda/common/codec.c @@ -606,9 +606,8 @@ void snd_hda_shutup_pins(struct hda_codec *codec) if (codec->bus->shutdown) return; snd_array_for_each(&codec->init_pins, i, pin) { - /* use read here for syncing after issuing each verb */ - snd_hda_codec_read(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + snd_hda_codec_write_sync(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); } codec->pins_shutup = 1; } @@ -2529,7 +2528,10 @@ EXPORT_SYMBOL_GPL(snd_hda_spdif_ctls_assign); static int spdif_share_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_multi_out *mout = (void *)kcontrol->private_value; + + guard(mutex)(&codec->spdif_mutex); ucontrol->value.integer.value[0] = mout->share_spdif; return 0; } @@ -2537,9 +2539,15 @@ static int spdif_share_sw_get(struct snd_kcontrol *kcontrol, static int spdif_share_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct hda_multi_out *mout = snd_kcontrol_chip(kcontrol); - mout->share_spdif = !!ucontrol->value.integer.value[0]; - return 0; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hda_multi_out *mout = (void *)kcontrol->private_value; + bool val = !!ucontrol->value.integer.value[0]; + int change; + + guard(mutex)(&codec->spdif_mutex); + change = mout->share_spdif != val; + mout->share_spdif = val; + return change; } static const struct snd_kcontrol_new spdif_share_sw = { @@ -2550,6 +2558,14 @@ static const struct snd_kcontrol_new spdif_share_sw = { .put = spdif_share_sw_put, }; +static void notify_spdif_share_sw(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + if (mout->share_spdif_kctl) + snd_ctl_notify_one(codec->card, SNDRV_CTL_EVENT_MASK_VALUE, + mout->share_spdif_kctl, 0); +} + /** * snd_hda_create_spdif_share_sw - create Default PCM switch * @codec: the HDA codec @@ -2559,15 +2575,24 @@ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { struct snd_kcontrol *kctl; + int err; if (!mout->dig_out_nid) return 0; - kctl = snd_ctl_new1(&spdif_share_sw, mout); + kctl = snd_ctl_new1(&spdif_share_sw, codec); if (!kctl) return -ENOMEM; - /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); + /* snd_ctl_new1() stores @codec in private_data; stash @mout in + * private_value for the share-switch callbacks and cache the + * assigned control for forced-disable notifications. + */ + kctl->private_value = (unsigned long)mout; + err = snd_hda_ctl_add(codec, mout->dig_out_nid, kctl); + if (err < 0) + return err; + mout->share_spdif_kctl = kctl; + return 0; } EXPORT_SYMBOL_GPL(snd_hda_create_spdif_share_sw); @@ -2768,9 +2793,9 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, if (codec->power_filter) state = codec->power_filter(codec, fg, state); if (state == power_state || power_state != AC_PWRST_D3) - snd_hda_codec_read(codec, fg, flags, - AC_VERB_SET_POWER_STATE, - state); + snd_hda_codec_write_sync(codec, fg, flags, + AC_VERB_SET_POWER_STATE, + state); snd_hda_codec_set_power_to_all(codec, fg, power_state); } state = snd_hda_sync_power_state(codec, fg, power_state); @@ -3701,6 +3726,8 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_pcm_stream *hinfo) { struct snd_pcm_runtime *runtime = substream->runtime; + bool notify_share_sw = false; + runtime->hw.channels_max = mout->max_channels; if (mout->dig_out_nid) { if (!mout->analog_rates) { @@ -3729,10 +3756,12 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, hinfo->maxbps = mout->spdif_maxbps; } else { mout->share_spdif = 0; - /* FIXME: need notify? */ + notify_share_sw = true; } } } + if (notify_share_sw) + notify_spdif_share_sw(codec, mout); return snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); } @@ -4024,6 +4053,35 @@ void snd_hda_bus_reset_codecs(struct hda_bus *bus) } /** + * snd_hda_codec_set_gpio - Set up GPIO bits for AFG + * @codec: the HDA codec + * @mask: GPIO bitmask + * @dir: GPIO direction bits + * @data: GPIO data bits + * @delay: the delay in msec before writing GPIO data bits + */ +void snd_hda_codec_set_gpio(struct hda_codec *codec, unsigned int mask, + unsigned int dir, unsigned int data, + unsigned int delay) +{ + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_MASK, mask); + if (delay) { + snd_hda_codec_write_sync(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DIRECTION, dir); + msleep(delay); + snd_hda_codec_write_sync(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, data); + } else { + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DIRECTION, dir); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, data); + } +} +EXPORT_SYMBOL_GPL(snd_hda_codec_set_gpio); + +/** * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer * @pcm: PCM caps bits * @buf: the string buffer to write diff --git a/sound/hda/common/hda_local.h b/sound/hda/common/hda_local.h index ab423f1cef54..98b2c4acebc2 100644 --- a/sound/hda/common/hda_local.h +++ b/sound/hda/common/hda_local.h @@ -221,6 +221,7 @@ struct hda_multi_out { unsigned int spdif_rates; unsigned int spdif_maxbps; u64 spdif_formats; + struct snd_kcontrol *share_spdif_kctl; /* cached shared SPDIF switch */ }; int snd_hda_create_spdif_share_sw(struct hda_codec *codec, diff --git a/sound/hda/common/proc.c b/sound/hda/common/proc.c index 3bc33c5617b2..c83796b13d3d 100644 --- a/sound/hda/common/proc.c +++ b/sound/hda/common/proc.c @@ -640,41 +640,78 @@ static void print_gpio(struct snd_info_buffer *buffer, { unsigned int gpio = param_read(codec, codec->core.afg, AC_PAR_GPIO_CAP); - unsigned int enable, direction, wake, unsol, sticky, data; - int i, max; + int i, gpio_max, gpo_max, gpi_max; + + gpio_max = gpio & AC_GPIO_IO_COUNT; + gpo_max = (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT; + gpi_max = (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT; + snd_iprintf(buffer, "GPIO: io=%d, o=%d, i=%d, " "unsolicited=%d, wake=%d\n", - gpio & AC_GPIO_IO_COUNT, - (gpio & AC_GPIO_O_COUNT) >> AC_GPIO_O_COUNT_SHIFT, - (gpio & AC_GPIO_I_COUNT) >> AC_GPIO_I_COUNT_SHIFT, + gpio_max, gpo_max, gpi_max, (gpio & AC_GPIO_UNSOLICITED) ? 1 : 0, (gpio & AC_GPIO_WAKE) ? 1 : 0); - max = gpio & AC_GPIO_IO_COUNT; - if (!max || max > 8) - return; - enable = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_MASK, 0); - direction = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - wake = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_WAKE_MASK, 0); - unsol = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, 0); - sticky = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_STICKY_MASK, 0); - data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DATA, 0); - for (i = 0; i < max; ++i) - snd_iprintf(buffer, - " IO[%d]: enable=%d, dir=%d, wake=%d, " - "sticky=%d, data=%d, unsol=%d\n", i, - (enable & (1<<i)) ? 1 : 0, - (direction & (1<<i)) ? 1 : 0, - (wake & (1<<i)) ? 1 : 0, - (sticky & (1<<i)) ? 1 : 0, - (data & (1<<i)) ? 1 : 0, - (unsol & (1<<i)) ? 1 : 0); - /* FIXME: add GPO and GPI pin information */ + + if (gpio_max && gpio_max <= 8) { + unsigned int enable, direction, wake, unsol, sticky, data; + + enable = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_MASK, 0); + direction = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + wake = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_WAKE_MASK, 0); + unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, + 0); + sticky = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_STICKY_MASK, 0); + data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPIO_DATA, 0); + for (i = 0; i < gpio_max; ++i) { + snd_iprintf(buffer, + " IO[%d]: enable=%d, dir=%d, wake=%d, ", + i, (enable & (1 << i)) ? 1 : 0, + (direction & (1 << i)) ? 1 : 0, + (wake & (1 << i)) ? 1 : 0); + snd_iprintf(buffer, + "sticky=%d, data=%d, unsol=%d\n", + (sticky & (1 << i)) ? 1 : 0, + (data & (1 << i)) ? 1 : 0, + (unsol & (1 << i)) ? 1 : 0); + } + } + + if (gpo_max && gpo_max <= 8) { + unsigned int gpo_data; + + gpo_data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPO_DATA, 0); + for (i = 0; i < gpo_max; ++i) + snd_iprintf(buffer, " GPO[%d]: data=%d\n", i, + (gpo_data & (1 << i)) ? 1 : 0); + } + + if (gpi_max && gpi_max <= 8) { + unsigned int wake, unsol, sticky, data; + + wake = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPI_WAKE_MASK, 0); + unsol = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPI_UNSOLICITED_RSP_MASK, + 0); + sticky = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPI_STICKY_MASK, 0); + data = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_GPI_DATA, 0); + for (i = 0; i < gpi_max; ++i) + snd_iprintf(buffer, " GPI[%d]: wake=%d, sticky=%d, data=%d, unsol=%d\n", + i, (wake & (1 << i)) ? 1 : 0, + (sticky & (1 << i)) ? 1 : 0, + (data & (1 << i)) ? 1 : 0, + (unsol & (1 << i)) ? 1 : 0); + } + print_nid_array(buffer, codec, nid, &codec->mixers); print_nid_array(buffer, codec, nid, &codec->nids); } @@ -940,4 +977,3 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) snprintf(name, sizeof(name), "codec#%d", codec->core.addr); return snd_card_ro_proc_new(codec->card, name, codec, print_codec_info); } - diff --git a/sound/hda/controllers/intel.c b/sound/hda/controllers/intel.c index a4104376523a..257c498c3260 100644 --- a/sound/hda/controllers/intel.c +++ b/sound/hda/controllers/intel.c @@ -2436,20 +2436,7 @@ static void azx_remove(struct pci_dev *pci) /* cancel the pending probing work */ chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - /* FIXME: below is an ugly workaround. - * Both device_release_driver() and driver_probe_device() - * take *both* the device's and its parent's lock before - * calling the remove() and probe() callbacks. The codec - * probe takes the locks of both the codec itself and its - * parent, i.e. the PCI controller dev. Meanwhile, when - * the PCI controller is unbound, it takes its lock, too - * ==> ouch, a deadlock! - * As a workaround, we unlock temporarily here the controller - * device during cancel_work_sync() call. - */ - device_unlock(&pci->dev); cancel_delayed_work_sync(&hda->probe_work); - device_lock(&pci->dev); clear_bit(chip->dev_index, probed_devs); pci_set_drvdata(pci, NULL); diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index b24c80410d45..9dd36b82a6ac 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -53,6 +53,31 @@ static void ak4524_reset(struct snd_akm4xxx *ak, int state) } } +/* reset procedure for AK4529 */ +static void ak4529_reset(struct snd_akm4xxx *ak, int state) +{ + static const unsigned char regs[] = { + 0x0a, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, + 0x06, 0x07, 0x0b, 0x0c, 0x08, + }; + unsigned int i; + unsigned char reg; + + if (state) { + snd_akm4xxx_write(ak, 0, 0x09, + snd_akm4xxx_get(ak, 0, 0x09) & ~0x01); + return; + } + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + reg = regs[i]; + snd_akm4xxx_write(ak, 0, reg, + snd_akm4xxx_get(ak, 0, reg)); + } + snd_akm4xxx_write(ak, 0, 0x09, + snd_akm4xxx_get(ak, 0, 0x09) | 0x01); +} + /* reset procedure for AK4355 and AK4358 */ static void ak435X_reset(struct snd_akm4xxx *ak, int state) { @@ -99,7 +124,7 @@ void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) ak4524_reset(ak, state); break; case SND_AK4529: - /* FIXME: needed for ak4529? */ + ak4529_reset(ak, state); break; case SND_AK4355: ak435X_reset(ak, state); @@ -256,6 +281,9 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) 0x07, 0x00, /* 7: ROUT muted */ 0xff, 0xff }; + static const unsigned char ak5365_defaults[] = { + 0x01, 0x00, 0x00, 0x2b, 0x7f, 0x7f, 0x28, 0x89, + }; int chip; const unsigned char *ptr, *inits; @@ -302,10 +330,12 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) ak->total_regs = 0x05; break; case SND_AK5365: - /* FIXME: any init sequence? */ ak->num_chips = 1; ak->name = "ak5365"; ak->total_regs = 0x08; + memcpy(ak->images, ak5365_defaults, sizeof(ak5365_defaults)); + snd_akm4xxx_set_vol(ak, 0, 0x04, 127); + snd_akm4xxx_set_vol(ak, 0, 0x05, 127); return; case SND_AK4620: inits = inits_ak4620; diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index e8c50a036bea..39c5e87c6ab0 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -51,6 +51,7 @@ int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer) snd_i2c_unlock(bus); return res; } +EXPORT_SYMBOL(snd_tea6330t_detect); #if 0 static void snd_tea6330t_set(struct tea6330t *tea, @@ -355,6 +356,42 @@ int snd_tea6330t_update_mixer(struct snd_card *card, snd_i2c_device_free(device); return err; } - -EXPORT_SYMBOL(snd_tea6330t_detect); EXPORT_SYMBOL(snd_tea6330t_update_mixer); + +int snd_tea6330t_restore_mixer(struct snd_i2c_bus *bus) +{ + struct snd_i2c_device *device; + struct tea6330t *tea; + unsigned char bytes[7]; + unsigned int idx; + int err; + + if (!bus) + return -EINVAL; + + snd_i2c_lock(bus); + list_for_each_entry(device, &bus->devices, list) { + if (device->addr != TEA6330T_ADDR) + continue; + + tea = device->private_data; + if (!tea) { + err = -EINVAL; + goto unlock; + } + + bytes[0] = TEA6330T_SADDR_VOLUME_LEFT; + for (idx = 0; idx < 6; idx++) + bytes[idx + 1] = tea->regs[idx]; + err = snd_i2c_sendbytes(device, bytes, 7); + err = err < 0 ? err : 0; + goto unlock; + } + + err = -ENODEV; + +unlock: + snd_i2c_unlock(bus); + return err; +} +EXPORT_SYMBOL(snd_tea6330t_restore_mixer); diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 6a95dfb7600a..7255b34f9148 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -184,12 +184,44 @@ static int snd_es1688_isa_probe(struct device *dev, unsigned int n) return 0; } +#ifdef CONFIG_PM +static int snd_es1688_card_suspend(struct snd_card *card) +{ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + return 0; +} + +static int snd_es1688_card_resume(struct snd_card *card) +{ + struct snd_es1688 *chip = card->private_data; + int err; + + err = snd_es1688_reset(chip); + if (err < 0) + return err; + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_es1688_isa_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_es1688_card_suspend(dev_get_drvdata(dev)); +} + +static int snd_es1688_isa_resume(struct device *dev, unsigned int n) +{ + return snd_es1688_card_resume(dev_get_drvdata(dev)); +} +#endif + static struct isa_driver snd_es1688_driver = { .match = snd_es1688_match, .probe = snd_es1688_isa_probe, -#if 0 /* FIXME */ - .suspend = snd_es1688_suspend, - .resume = snd_es1688_resume, +#ifdef CONFIG_PM + .suspend = snd_es1688_isa_suspend, + .resume = snd_es1688_isa_resume, #endif .driver = { .name = DEV_NAME @@ -266,20 +298,12 @@ static void snd_es968_pnp_remove(struct pnp_card_link *pcard) static int snd_es968_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) { - struct snd_card *card = pnp_get_card_drvdata(pcard); - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - return 0; + return snd_es1688_card_suspend(pnp_get_card_drvdata(pcard)); } static int snd_es968_pnp_resume(struct pnp_card_link *pcard) { - struct snd_card *card = pnp_get_card_drvdata(pcard); - struct snd_es1688 *chip = card->private_data; - - snd_es1688_reset(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; + return snd_es1688_card_resume(pnp_get_card_drvdata(pcard)); } #endif diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index ffc69e26227e..30bd76eee96e 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -173,6 +173,39 @@ int snd_gf1_dma_done(struct snd_gus_card * gus) return 0; } +void snd_gf1_dma_suspend(struct snd_gus_card *gus) +{ + struct snd_gf1_dma_block *block; + + guard(mutex)(&gus->dma_mutex); + if (!gus->gf1.dma_shared) + return; + + snd_dma_disable(gus->gf1.dma1); + snd_gf1_dma_ack(gus); + if (gus->gf1.dma_ack) + gus->gf1.dma_ack(gus, gus->gf1.dma_private_data); + gus->gf1.dma_ack = NULL; + gus->gf1.dma_private_data = NULL; + + while ((block = gus->gf1.dma_data_pcm)) { + gus->gf1.dma_data_pcm = block->next; + if (block->ack) + block->ack(gus, block->private_data); + kfree(block); + } + while ((block = gus->gf1.dma_data_synth)) { + gus->gf1.dma_data_synth = block->next; + if (block->ack) + block->ack(gus, block->private_data); + kfree(block); + } + + gus->gf1.dma_data_pcm_last = NULL; + gus->gf1.dma_data_synth_last = NULL; + gus->gf1.dma_flags &= ~SNDRV_GF1_DMA_TRIGGER; +} + int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, struct snd_gf1_dma_block * __block, int atomic, diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index b2b189c83569..6adf8b698e2b 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -404,6 +404,42 @@ int snd_gus_initialize(struct snd_gus_card *gus) return 0; } +int snd_gus_suspend(struct snd_gus_card *gus) +{ + int err; + + if (gus->pcm) { + err = snd_pcm_suspend_all(gus->pcm); + if (err < 0) + return err; + } + + err = snd_gf1_suspend(gus); + if (err < 0) + return err; + + snd_power_change_state(gus->card, SNDRV_CTL_POWER_D3hot); + return 0; +} +EXPORT_SYMBOL(snd_gus_suspend); + +int snd_gus_resume(struct snd_gus_card *gus) +{ + int err; + + err = snd_gus_init_dma_irq(gus, 1); + if (err < 0) + return err; + + err = snd_gf1_resume(gus); + if (err < 0) + return err; + + snd_power_change_state(gus->card, SNDRV_CTL_POWER_D0); + return 0; +} +EXPORT_SYMBOL(snd_gus_resume); + /* gus_io.c */ EXPORT_SYMBOL(snd_gf1_delay); EXPORT_SYMBOL(snd_gf1_write8); diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index caf371897b78..a0757e1ede46 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -471,7 +471,8 @@ static int snd_gf1_pcm_playback_trigger(struct snd_pcm_substream *substream, if (cmd == SNDRV_PCM_TRIGGER_START) { snd_gf1_pcm_trigger_up(substream); - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + } else if (cmd == SNDRV_PCM_TRIGGER_STOP || + cmd == SNDRV_PCM_TRIGGER_SUSPEND) { scoped_guard(spinlock, &pcmp->lock) { pcmp->flags &= ~SNDRV_GF1_PCM_PFLG_ACTIVE; } @@ -558,7 +559,8 @@ static int snd_gf1_pcm_capture_trigger(struct snd_pcm_substream *substream, if (cmd == SNDRV_PCM_TRIGGER_START) { val = gus->gf1.pcm_rcntrl_reg; - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + } else if (cmd == SNDRV_PCM_TRIGGER_STOP || + cmd == SNDRV_PCM_TRIGGER_SUSPEND) { val = 0; } else { return -EINVAL; @@ -856,4 +858,3 @@ int snd_gf1_pcm_new(struct snd_gus_card *gus, int pcm_dev, int control_index) return 0; } - diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index a7a3e764bb77..998fa245708c 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -6,6 +6,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/time.h> +#include <asm/dma.h> #include <sound/core.h> #include <sound/gus.h> @@ -263,11 +264,18 @@ void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice) private_free(voice); } -/* - * call this function only by start of driver - */ +static void snd_gf1_init_software_state(struct snd_gus_card *gus) +{ + unsigned int i; -int snd_gf1_start(struct snd_gus_card * gus) + snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_ALL); + for (i = 0; i < 32; i++) { + gus->gf1.voices[i].number = i; + snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_VOICE | i); + } +} + +static void snd_gf1_hw_start(struct snd_gus_card *gus, bool initial) { unsigned int i; @@ -277,14 +285,14 @@ int snd_gf1_start(struct snd_gus_card * gus) udelay(160); snd_gf1_i_write8(gus, SNDRV_GF1_GB_JOYSTICK_DAC_LEVEL, gus->joystick_dac); - snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_ALL); - for (i = 0; i < 32; i++) { - gus->gf1.voices[i].number = i; - snd_gf1_set_default_handlers(gus, SNDRV_GF1_HANDLER_VOICE | i); + if (initial) { + snd_gf1_init_software_state(gus); + snd_gf1_uart_cmd(gus, 0x03); + } else { + guard(spinlock_irqsave)(&gus->uart_cmd_lock); + outb(0x03, GUSP(gus, MIDICTRL)); } - snd_gf1_uart_cmd(gus, 0x03); /* huh.. this cleanup took me some time... */ - if (gus->gf1.enh_mode) { /* enhanced mode !!!! */ snd_gf1_i_write8(gus, SNDRV_GF1_GB_GLOBAL_MODE, snd_gf1_i_look8(gus, SNDRV_GF1_GB_GLOBAL_MODE) | 0x01); snd_gf1_i_write8(gus, SNDRV_GF1_GB_MEMORY_CONTROL, 0x01); @@ -293,6 +301,8 @@ int snd_gf1_start(struct snd_gus_card * gus) snd_gf1_select_active_voices(gus); snd_gf1_delay(gus); gus->gf1.default_voice_address = gus->gf1.memory > 0 ? 0 : 512 - 8; + gus->gf1.hw_lfo = 0; + gus->gf1.sw_lfo = 0; /* initialize LFOs & clear LFOs memory */ if (gus->gf1.enh_mode && gus->gf1.memory) { gus->gf1.hw_lfo = 1; @@ -321,7 +331,15 @@ int snd_gf1_start(struct snd_gus_card * gus) outb(gus->gf1.active_voice = 0, GUSP(gus, GF1PAGE)); outb(gus->mix_cntrl_reg, GUSP(gus, MIXCNTRLREG)); } +} +int snd_gf1_start(struct snd_gus_card *gus) +{ + /* + * Probe-time startup initializes both GF1 hardware and the + * software state that suspend/resume keeps across PM cycles. + */ + snd_gf1_hw_start(gus, true); snd_gf1_timers_init(gus); snd_gf1_look_regs(gus); snd_gf1_mem_init(gus); @@ -357,3 +375,27 @@ int snd_gf1_stop(struct snd_gus_card * gus) return 0; } + +int snd_gf1_suspend(struct snd_gus_card *gus) +{ + snd_gf1_dma_suspend(gus); + snd_gf1_uart_suspend(gus); + + snd_gf1_i_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, 0); + snd_gf1_i_write8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL, 0); + snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL); + snd_gf1_stop_voices(gus, 0, 31); + snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 1); + snd_dma_disable(gus->gf1.dma2); + + return 0; +} + +int snd_gf1_resume(struct snd_gus_card *gus) +{ + snd_gf1_hw_start(gus, false); + snd_gf1_timers_resume(gus); + snd_gf1_uart_resume(gus); + + return 0; +} diff --git a/sound/isa/gus/gus_timer.c b/sound/isa/gus/gus_timer.c index e3a8847e02cf..14dcde138bc7 100644 --- a/sound/isa/gus/gus_timer.c +++ b/sound/isa/gus/gus_timer.c @@ -178,3 +178,17 @@ void snd_gf1_timers_done(struct snd_gus_card * gus) gus->gf1.timer2 = NULL; } } + +void snd_gf1_timers_resume(struct snd_gus_card *gus) +{ + if (gus->gf1.timer1) { + gus->gf1.interrupt_handler_timer1 = snd_gf1_interrupt_timer1; + if (gus->gf1.timer_enabled & 4) + snd_gf1_timer1_start(gus->gf1.timer1); + } + if (gus->gf1.timer2) { + gus->gf1.interrupt_handler_timer2 = snd_gf1_interrupt_timer2; + if (gus->gf1.timer_enabled & 8) + snd_gf1_timer2_start(gus->gf1.timer2); + } +} diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index 770d8f3e4cff..25057a5a81b0 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -232,3 +232,50 @@ int snd_gf1_rawmidi_new(struct snd_gus_card *gus, int device) gus->midi_uart = rmidi; return err; } + +void snd_gf1_uart_suspend(struct snd_gus_card *gus) +{ + guard(spinlock_irqsave)(&gus->uart_cmd_lock); + outb(0x03, GUSP(gus, MIDICTRL)); +} + +void snd_gf1_uart_resume(struct snd_gus_card *gus) +{ + unsigned short uart_cmd; + bool active; + int i; + + scoped_guard(spinlock_irqsave, &gus->uart_cmd_lock) { + active = gus->midi_substream_input || gus->midi_substream_output; + } + if (!active) + return; + + /* snd_gf1_hw_start() already left MIDICTRL in reset. */ + usleep_range(160, 200); + + guard(spinlock_irqsave)(&gus->uart_cmd_lock); + if (!gus->midi_substream_input && !gus->midi_substream_output) + return; + + if (gus->midi_substream_output) + gus->gf1.interrupt_handler_midi_out = snd_gf1_interrupt_midi_out; + if (gus->midi_substream_input) + gus->gf1.interrupt_handler_midi_in = snd_gf1_interrupt_midi_in; + + if (!gus->uart_enable) + return; + + uart_cmd = gus->gf1.uart_cmd; + snd_gf1_uart_cmd(gus, 0x00); + + if (gus->midi_substream_input) { + for (i = 0; i < 1000 && (snd_gf1_uart_stat(gus) & 0x01); i++) + snd_gf1_uart_get(gus); + if (i >= 1000) + dev_err(gus->card->dev, + "gus midi uart resume - cleanup error\n"); + } + + snd_gf1_uart_cmd(gus, uart_cmd); +} diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 101202acefb3..363c819ced89 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -145,6 +145,7 @@ static int snd_gusclassic_probe(struct device *dev, unsigned int n) error = snd_gusclassic_create(card, dev, n, &gus); if (error < 0) return error; + card->private_data = gus; error = snd_gusclassic_detect(gus); if (error < 0) @@ -193,11 +194,29 @@ static int snd_gusclassic_probe(struct device *dev, unsigned int n) return 0; } +#ifdef CONFIG_PM +static int snd_gusclassic_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(dev); + + return snd_gus_suspend(card->private_data); +} + +static int snd_gusclassic_resume(struct device *dev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(dev); + + return snd_gus_resume(card->private_data); +} +#endif + static struct isa_driver snd_gusclassic_driver = { .match = snd_gusclassic_match, .probe = snd_gusclassic_probe, -#if 0 /* FIXME */ +#ifdef CONFIG_PM .suspend = snd_gusclassic_suspend, + .resume = snd_gusclassic_resume, #endif .driver = { .name = DEV_NAME diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index ed921b89b00a..0984731740c4 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -44,6 +44,11 @@ static int joystick_dac[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 29}; static int channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 24}; static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; +struct snd_gusextreme { + struct snd_es1688 es1688; + struct snd_gus_card *gus; +}; + module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); module_param_array(id, charp, NULL, 0444); @@ -142,17 +147,15 @@ static int snd_gusextreme_gus_card_create(struct snd_card *card, 0, channels[n], pcm_channels[n], 0, rgus); } -static int snd_gusextreme_detect(struct snd_gus_card *gus, - struct snd_es1688 *es1688) +static void snd_gusextreme_enable_gf1(struct snd_gus_card *gus, + struct snd_es1688 *es1688) { - unsigned char d; - /* * This is main stuff - enable access to GF1 chip... * I'm not sure, if this will work for card which have * ES1688 chip in another place than 0x220. - * - * I used reverse-engineering in DOSEMU. [--jk] + * + * I used reverse-engineering in DOSEMU. [--jk] * * ULTRINIT.EXE: * 0x230 = 0,2,3 @@ -172,7 +175,14 @@ static int snd_gusextreme_detect(struct snd_gus_card *gus, outb(0, 0x201); outb(gus->gf1.port & 0x010 ? 3 : 1, ES1688P(es1688, INIT1)); } +} + +static int snd_gusextreme_detect(struct snd_gus_card *gus, + struct snd_es1688 *es1688) +{ + unsigned char d; + snd_gusextreme_enable_gf1(gus, es1688); udelay(100); snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */ @@ -223,16 +233,18 @@ static int snd_gusextreme_probe(struct device *dev, unsigned int n) { struct snd_card *card; struct snd_gus_card *gus; + struct snd_gusextreme *gusextreme; struct snd_es1688 *es1688; struct snd_opl3 *opl3; int error; error = snd_devm_card_new(dev, index[n], id[n], THIS_MODULE, - sizeof(struct snd_es1688), &card); + sizeof(*gusextreme), &card); if (error < 0) return error; - es1688 = card->private_data; + gusextreme = card->private_data; + es1688 = &gusextreme->es1688; if (mpu_port[n] == SNDRV_AUTO_PORT) mpu_port[n] = 0; @@ -250,6 +262,7 @@ static int snd_gusextreme_probe(struct device *dev, unsigned int n) error = snd_gusextreme_gus_card_create(card, dev, n, &gus); if (error < 0) return error; + gusextreme->gus = gus; error = snd_gusextreme_detect(gus, es1688); if (error < 0) @@ -321,10 +334,36 @@ static int snd_gusextreme_probe(struct device *dev, unsigned int n) return 0; } +#ifdef CONFIG_PM +static int snd_gusextreme_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_gusextreme *gusextreme = card->private_data; + + return snd_gus_suspend(gusextreme->gus); +} + +static int snd_gusextreme_resume(struct device *dev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_gusextreme *gusextreme = card->private_data; + int err; + + err = snd_es1688_reset(&gusextreme->es1688); + if (err < 0) + return err; + + snd_gusextreme_enable_gf1(gusextreme->gus, &gusextreme->es1688); + usleep_range(100, 200); + return snd_gus_resume(gusextreme->gus); +} +#endif + static struct isa_driver snd_gusextreme_driver = { .match = snd_gusextreme_match, .probe = snd_gusextreme_probe, -#if 0 /* FIXME */ +#ifdef CONFIG_PM .suspend = snd_gusextreme_suspend, .resume = snd_gusextreme_resume, #endif diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index b572411c4422..f1fd7ff2121d 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -328,12 +328,38 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) return 0; } +#ifdef CONFIG_PM +static int snd_gusmax_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_gusmax *maxcard = card->private_data; + + maxcard->wss->suspend(maxcard->wss); + return snd_gus_suspend(maxcard->gus); +} + +static int snd_gusmax_resume(struct device *dev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_gusmax *maxcard = card->private_data; + + /* Restore the board routing latch before resuming the codec and GF1. */ + outb(maxcard->gus->max_cntrl_val, GUSP(maxcard->gus, MAXCNTRLPORT)); + maxcard->wss->resume(maxcard->wss); + return snd_gus_resume(maxcard->gus); +} +#endif + #define DEV_NAME "gusmax" static struct isa_driver snd_gusmax_driver = { .match = snd_gusmax_match, .probe = snd_gusmax_probe, - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_gusmax_suspend, + .resume = snd_gusmax_resume, +#endif .driver = { .name = DEV_NAME }, diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 18adcd35e117..6c3a2977dcb3 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -96,6 +96,7 @@ struct snd_interwave { struct snd_gus_card *gus; struct snd_wss *wss; #ifdef SNDRV_STB + struct snd_i2c_bus *i2c_bus; struct resource *i2c_res; #endif unsigned short gus_status_reg; @@ -363,18 +364,30 @@ struct rom_hdr { /* 511 */ unsigned char csum; }; -static void snd_interwave_detect_memory(struct snd_gus_card *gus) +static const unsigned int snd_interwave_memory_configs[] = { + 0x00000001, 0x00000101, 0x01010101, 0x00000401, + 0x04040401, 0x00040101, 0x04040101, 0x00000004, + 0x00000404, 0x04040404, 0x00000010, 0x00001010, + 0x10101010 +}; + +static int snd_interwave_find_memory_config(unsigned int lmct) { - static const unsigned int lmc[13] = - { - 0x00000001, 0x00000101, 0x01010101, 0x00000401, - 0x04040401, 0x00040101, 0x04040101, 0x00000004, - 0x00000404, 0x04040404, 0x00000010, 0x00001010, - 0x10101010 - }; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(snd_interwave_memory_configs); i++) { + if (lmct == snd_interwave_memory_configs[i]) + return i; + } + + return -EINVAL; +} +static void snd_interwave_detect_memory(struct snd_gus_card *gus) +{ int bank_pos, pages; unsigned int i, lmct; + int lmc_cfg; int psizes[4]; unsigned char iwave[8]; unsigned char csum; @@ -399,17 +412,20 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) #if 0 dev_dbg(gus->card->dev, "lmct = 0x%08x\n", lmct); #endif - for (i = 0; i < ARRAY_SIZE(lmc); i++) - if (lmct == lmc[i]) { + lmc_cfg = snd_interwave_find_memory_config(lmct); + if (lmc_cfg >= 0) { #if 0 - dev_dbg(gus->card->dev, "found !!! %i\n", i); + dev_dbg(gus->card->dev, "found !!! %i\n", lmc_cfg); #endif - snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i); - snd_interwave_bank_sizes(gus, psizes); - break; - } - if (i >= ARRAY_SIZE(lmc) && !gus->gf1.enh_mode) - snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | 2); + snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, + (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | + lmc_cfg); + snd_interwave_bank_sizes(gus, psizes); + } else if (!gus->gf1.enh_mode) { + snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, + (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | + 2); + } for (i = 0; i < 4; i++) { gus->gf1.mem_alloc.banks_8[i].address = gus->gf1.mem_alloc.banks_16[i].address = i << 22; @@ -454,24 +470,28 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) snd_interwave_reset(gus); } +static void __snd_interwave_restore_regs(struct snd_gus_card *gus) +{ + snd_gf1_write8(gus, SNDRV_GF1_GB_COMPATIBILITY, 0x1f); + snd_gf1_write8(gus, SNDRV_GF1_GB_DECODE_CONTROL, 0x49); + snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, 0x11); + snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A, 0x00); + snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B, 0x30); + snd_gf1_write8(gus, SNDRV_GF1_GB_EMULATION_IRQ, 0x00); +} + static void snd_interwave_init(int dev, struct snd_gus_card *gus) { - /* ok.. some InterWave specific initialization */ + /* Probe-time setup also clears the timer control register. */ scoped_guard(spinlock_irqsave, &gus->reg_lock) { snd_gf1_write8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL, 0x00); - snd_gf1_write8(gus, SNDRV_GF1_GB_COMPATIBILITY, 0x1f); - snd_gf1_write8(gus, SNDRV_GF1_GB_DECODE_CONTROL, 0x49); - snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, 0x11); - snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A, 0x00); - snd_gf1_write8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B, 0x30); - snd_gf1_write8(gus, SNDRV_GF1_GB_EMULATION_IRQ, 0x00); + __snd_interwave_restore_regs(gus); } gus->equal_irq = 1; gus->codec_flag = 1; gus->interwave = 1; gus->max_flag = 1; gus->joystick_dac = joystick_dac[dev]; - } static const struct snd_kcontrol_new snd_interwave_controls[] = { @@ -724,6 +744,7 @@ static int snd_interwave_probe(struct snd_card *card, int dev, err = snd_tea6330t_update_mixer(card, i2c_bus, 0, 1); if (err < 0) return err; + iwcard->i2c_bus = i2c_bus; } #endif @@ -828,10 +849,97 @@ static int snd_interwave_isa_probe(struct device *pdev, return 0; } +#ifdef CONFIG_PM +static void snd_interwave_restore_regs(struct snd_gus_card *gus) +{ + scoped_guard(spinlock_irqsave, &gus->reg_lock) + __snd_interwave_restore_regs(gus); +} + +static void snd_interwave_restore_memory(struct snd_gus_card *gus) +{ + unsigned short mem_cfg; + unsigned int lmct = 0; + int i, lmc_cfg; + + if (!gus->gf1.memory) + return; + + for (i = 0; i < 4; i++) + lmct |= (gus->gf1.mem_alloc.banks_16[i].size >> 18) << (i * 8); + + lmc_cfg = snd_interwave_find_memory_config(lmct); + if (lmc_cfg < 0) { + if (!gus->gf1.enh_mode) { + lmc_cfg = 2; + } else { + dev_warn(gus->card->dev, + "cannot restore InterWave memory layout 0x%08x\n", + lmct); + return; + } + } + + scoped_guard(spinlock_irqsave, &gus->reg_lock) { + mem_cfg = snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG); + mem_cfg = (mem_cfg & 0xfff0) | lmc_cfg; + mem_cfg = (mem_cfg & 0xff1f) | (4 << 5); + snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, mem_cfg); + } +} + +static int snd_interwave_card_suspend(struct snd_card *card) +{ + struct snd_interwave *iwcard = card->private_data; + + iwcard->wss->suspend(iwcard->wss); + return snd_gus_suspend(iwcard->gus); +} + +static int snd_interwave_card_resume(struct snd_card *card) +{ + struct snd_interwave *iwcard = card->private_data; + int err; + + err = snd_gus_resume(iwcard->gus); + if (err < 0) + return err; + + snd_interwave_restore_regs(iwcard->gus); + snd_interwave_restore_memory(iwcard->gus); + iwcard->wss->resume(iwcard->wss); +#ifdef SNDRV_STB + if (iwcard->i2c_bus) { + err = snd_tea6330t_restore_mixer(iwcard->i2c_bus); + if (err < 0) + dev_warn(card->dev, + "failed to restore TEA6330T mixer state: %d\n", + err); + } +#endif + + return 0; +} + +static int snd_interwave_isa_suspend(struct device *pdev, unsigned int dev, + pm_message_t state) +{ + return snd_interwave_card_suspend(dev_get_drvdata(pdev)); +} + +static int snd_interwave_isa_resume(struct device *pdev, unsigned int dev) +{ + return snd_interwave_card_resume(dev_get_drvdata(pdev)); +} +#endif + static struct isa_driver snd_interwave_driver = { .match = snd_interwave_isa_match, .probe = snd_interwave_isa_probe, - /* FIXME: suspend,resume */ +#ifdef CONFIG_PM + .suspend = snd_interwave_isa_suspend, + .resume = snd_interwave_isa_resume, +#endif .driver = { .name = INTERWAVE_DRIVER }, @@ -871,12 +979,28 @@ static int snd_interwave_pnp_detect(struct pnp_card_link *pcard, return 0; } +#ifdef CONFIG_PM +static int snd_interwave_pnpc_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + return snd_interwave_card_suspend(pnp_get_card_drvdata(pcard)); +} + +static int snd_interwave_pnpc_resume(struct pnp_card_link *pcard) +{ + return snd_interwave_card_resume(pnp_get_card_drvdata(pcard)); +} +#endif + static struct pnp_card_driver interwave_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = INTERWAVE_PNP_DRIVER, .id_table = snd_interwave_pnpids, .probe = snd_interwave_pnp_detect, - /* FIXME: suspend,resume */ +#ifdef CONFIG_PM + .suspend = snd_interwave_pnpc_suspend, + .resume = snd_interwave_pnpc_resume, +#endif }; #endif /* CONFIG_PNP */ diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 5e350234d572..77367e102fda 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -127,11 +127,8 @@ int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) } EXPORT_SYMBOL(snd_msnd_upload_host); -int snd_msnd_enable_irq(struct snd_msnd *dev) +static int __snd_msnd_enable_irq(struct snd_msnd *dev) { - if (dev->irq_ref++) - return 0; - dev_dbg(dev->card->dev, LOGNAME ": Enabling IRQ\n"); guard(spinlock_irqsave)(&dev->lock); @@ -152,17 +149,9 @@ int snd_msnd_enable_irq(struct snd_msnd *dev) return -EIO; } -EXPORT_SYMBOL(snd_msnd_enable_irq); -int snd_msnd_disable_irq(struct snd_msnd *dev) +static int __snd_msnd_disable_irq(struct snd_msnd *dev) { - if (--dev->irq_ref > 0) - return 0; - - if (dev->irq_ref < 0) - dev_dbg(dev->card->dev, LOGNAME ": IRQ ref count is %d\n", - dev->irq_ref); - dev_dbg(dev->card->dev, LOGNAME ": Disabling IRQ\n"); guard(spinlock_irqsave)(&dev->lock); @@ -178,8 +167,39 @@ int snd_msnd_disable_irq(struct snd_msnd *dev) return -EIO; } + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + if (dev->irq_ref++) + return 0; + + return __snd_msnd_enable_irq(dev); +} +EXPORT_SYMBOL(snd_msnd_enable_irq); + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + dev_dbg(dev->card->dev, LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); + + return __snd_msnd_disable_irq(dev); +} EXPORT_SYMBOL(snd_msnd_disable_irq); +int snd_msnd_force_irq(struct snd_msnd *dev, bool enable) +{ + if (!dev->irq_ref) + return 0; + + return enable ? __snd_msnd_enable_irq(dev) : + __snd_msnd_disable_irq(dev); +} +EXPORT_SYMBOL(snd_msnd_force_irq); + static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) { long tmp = (size * HZ * chip->play_sample_size) / 8; @@ -507,25 +527,27 @@ static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_msnd *chip = snd_pcm_substream_chip(substream); - int result = 0; - if (cmd == SNDRV_PCM_TRIGGER_START) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: dev_dbg(chip->card->dev, "%s(START)\n", __func__); chip->banksPlayed = 0; set_bit(F_WRITING, &chip->flags); snd_msnd_DAPQ(chip, 1); - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(chip->card->dev, "%s(STOP)\n", __func__); - /* interrupt diagnostic, comment this out later */ clear_bit(F_WRITING, &chip->flags); snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); - } else { + break; + default: dev_dbg(chip->card->dev, "%s(?????)\n", __func__); - result = -EINVAL; + return -EINVAL; } dev_dbg(chip->card->dev, "%s() ENDE\n", __func__); - return result; + return 0; } static snd_pcm_uframes_t @@ -589,17 +611,22 @@ static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, { struct snd_msnd *chip = snd_pcm_substream_chip(substream); - if (cmd == SNDRV_PCM_TRIGGER_START) { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: chip->last_recbank = -1; set_bit(F_READING, &chip->flags); if (snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_START) == 0) return 0; clear_bit(F_READING, &chip->flags); - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: clear_bit(F_READING, &chip->flags); snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); return 0; + default: + break; } return -EINVAL; } @@ -668,4 +695,3 @@ EXPORT_SYMBOL(snd_msnd_pcm); MODULE_DESCRIPTION("Common routines for Turtle Beach Multisound drivers"); MODULE_LICENSE("GPL"); - diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h index 3d7810ed9186..56a700e6a5cb 100644 --- a/sound/isa/msnd/msnd.h +++ b/sound/isa/msnd/msnd.h @@ -253,6 +253,8 @@ struct snd_msnd { spinlock_t mixer_lock; int nresets; unsigned recsrc; + u8 pm_recsrc; + bool pm_mpu_input; #define LEVEL_ENTRIES 32 int left_levels[LEVEL_ENTRIES]; int right_levels[LEVEL_ENTRIES]; @@ -280,6 +282,7 @@ int snd_msnd_upload_host(struct snd_msnd *chip, const u8 *bin, int len); int snd_msnd_enable_irq(struct snd_msnd *chip); int snd_msnd_disable_irq(struct snd_msnd *chip); +int snd_msnd_force_irq(struct snd_msnd *chip, bool enable); void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file); int snd_msnd_DAPQ(struct snd_msnd *chip, int start); int snd_msnd_DARQ(struct snd_msnd *chip, int start); diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index c4eec391cd29..5b729bb02ef6 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -513,6 +513,19 @@ static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) snd_msnd_disable_irq(mpu->private_data); } +#ifdef CONFIG_PM +static u8 snd_msnd_pm_recsrc(struct snd_msnd *chip) +{ + /* Convert recsrc to the Capture Source selector: 0=Analog, 1=MASS, 2=SPDIF. */ + if (chip->recsrc & BIT(4)) + return 1; + if ((chip->recsrc & BIT(17)) && + test_bit(F_HAVEDIGITAL, &chip->flags)) + return 2; + return 0; +} +#endif + static long mpu_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; @@ -1001,10 +1014,73 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) return 0; } +#ifdef CONFIG_PM +static int snd_msnd_card_suspend(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + struct snd_mpu401 *mpu; + int err; + + mpu = chip->rmidi ? chip->rmidi->private_data : NULL; + chip->pm_recsrc = snd_msnd_pm_recsrc(chip); + chip->pm_mpu_input = mpu && test_bit(MPU401_MODE_BIT_INPUT, &mpu->mode); + if (chip->pm_mpu_input) + snd_msnd_send_dsp_cmd(chip, HDEX_MIDI_IN_STOP); + + err = snd_msnd_force_irq(chip, false); + if (err < 0) { + if (chip->pm_mpu_input) + snd_msnd_send_dsp_cmd(chip, HDEX_MIDI_IN_START); + return err; + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + return 0; +} + +static int snd_msnd_card_resume(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + + err = snd_msnd_initialize(card); + if (err < 0) + return err; + + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, chip->pm_recsrc); + + err = snd_msnd_force_irq(chip, true); + if (err < 0) + return err; + + if (chip->pm_mpu_input) + snd_msnd_send_dsp_cmd(chip, HDEX_MIDI_IN_START); + + chip->nresets = 0; + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_msnd_isa_suspend(struct device *dev, unsigned int idx, + pm_message_t state) +{ + return snd_msnd_card_suspend(dev_get_drvdata(dev)); +} + +static int snd_msnd_isa_resume(struct device *dev, unsigned int idx) +{ + return snd_msnd_card_resume(dev_get_drvdata(dev)); +} +#endif + static struct isa_driver snd_msnd_driver = { .match = snd_msnd_isa_match, .probe = snd_msnd_isa_probe, - /* FIXME: suspend, resume */ +#ifdef CONFIG_PM + .suspend = snd_msnd_isa_suspend, + .resume = snd_msnd_isa_resume, +#endif .driver = { .name = DEV_NAME }, @@ -1111,6 +1187,18 @@ static int snd_msnd_pnp_detect(struct pnp_card_link *pcard, return 0; } +#ifdef CONFIG_PM +static int snd_msnd_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) +{ + return snd_msnd_card_suspend(pnp_get_card_drvdata(pcard)); +} + +static int snd_msnd_pnp_resume(struct pnp_card_link *pcard) +{ + return snd_msnd_card_resume(pnp_get_card_drvdata(pcard)); +} +#endif + static int isa_registered; static int pnp_registered; @@ -1127,6 +1215,10 @@ static struct pnp_card_driver msnd_pnpc_driver = { .name = "msnd_pinnacle", .id_table = msnd_pnpids, .probe = snd_msnd_pnp_detect, +#ifdef CONFIG_PM + .suspend = snd_msnd_pnp_suspend, + .resume = snd_msnd_pnp_resume, +#endif }; #endif /* CONFIG_PNP */ @@ -1161,4 +1253,3 @@ static void __exit snd_msnd_exit(void) module_init(snd_msnd_init); module_exit(snd_msnd_exit); - diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index ec354483b9f8..8ca987221753 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -310,6 +310,10 @@ EXPORT_SYMBOL(snd_msndmix_new); void snd_msndmix_setup(struct snd_msnd *dev) { + writew(dev->left_levels[MSND_MIXER_VOLUME], + dev->SMA + SMA_wCurrMastVolLeft); + writew(dev->right_levels[MSND_MIXER_VOLUME], + dev->SMA + SMA_wCurrMastVolRight); update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); update_volm(MSND_MIXER_PCM, wCurrPlayVol); diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 6d618cc2ba45..cd3a63c7c2a7 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -100,6 +100,16 @@ MODULE_PARM_DESC(joystick, "Enable gameport."); #define PFX "sc6000: " #define DRV_NAME "SC-6000" +struct snd_sc6000 { + char __iomem *vport; + char __iomem *vmss_port; + struct snd_wss *chip; + u8 mss_config; + u8 config; + u8 hw_cfg[2]; + bool old_dsp; +}; + /* hardware dependent functions */ /* @@ -267,7 +277,7 @@ static int sc6000_dsp_reset(char __iomem *vport) /* detection and initialization */ static int sc6000_hw_cfg_write(struct device *devptr, - char __iomem *vport, const int *cfg) + char __iomem *vport, const u8 *cfg) { if (sc6000_write(devptr, vport, COMMAND_6C) < 0) { dev_warn(devptr, "CMD 0x%x: failed!\n", COMMAND_6C); @@ -353,8 +363,7 @@ static int sc6000_init_mss(struct device *devptr, return 0; } -static void sc6000_hw_cfg_encode(struct device *devptr, - char __iomem *vport, int *cfg, +static void sc6000_hw_cfg_encode(struct device *devptr, u8 *cfg, long xport, long xmpu, long xmss_port, int joystick) { @@ -376,27 +385,83 @@ static void sc6000_hw_cfg_encode(struct device *devptr, dev_dbg(devptr, "hw cfg %x, %x\n", cfg[0], cfg[1]); } -static int sc6000_init_board(struct device *devptr, - char __iomem *vport, - char __iomem *vmss_port, int dev) +static void sc6000_prepare_board(struct device *devptr, + struct snd_sc6000 *sc6000, + unsigned int dev, int xirq, int xdma) +{ + sc6000->mss_config = sc6000_irq_to_softcfg(xirq) | + sc6000_dma_to_softcfg(xdma); + sc6000->config = sc6000->mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); + sc6000_hw_cfg_encode(devptr, sc6000->hw_cfg, port[dev], mpu_port[dev], + mss_port[dev], joystick[dev]); +} + +static void sc6000_detect_old_dsp(struct device *devptr, + struct snd_sc6000 *sc6000) +{ + sc6000_write(devptr, sc6000->vport, COMMAND_5C); + sc6000->old_dsp = sc6000_read(sc6000->vport) < 0; +} + +static int sc6000_program_board(struct device *devptr, + struct snd_sc6000 *sc6000) +{ + int err; + + if (!sc6000->old_dsp) { + if (sc6000_hw_cfg_write(devptr, sc6000->vport, + sc6000->hw_cfg) < 0) { + dev_err(devptr, "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } + } + + err = sc6000_setup_board(devptr, sc6000->vport, sc6000->config); + if (err < 0) { + dev_err(devptr, "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + sc6000_dsp_reset(sc6000->vport); + + if (!sc6000->old_dsp) { + sc6000_write(devptr, sc6000->vport, COMMAND_60); + sc6000_write(devptr, sc6000->vport, 0x02); + sc6000_dsp_reset(sc6000->vport); + } + + err = sc6000_setup_board(devptr, sc6000->vport, sc6000->config); + if (err < 0) { + dev_err(devptr, "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + err = sc6000_init_mss(devptr, sc6000->vport, sc6000->config, + sc6000->vmss_port, sc6000->mss_config); + if (err < 0) { + dev_err(devptr, "Cannot initialize Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int sc6000_init_board(struct device *devptr, struct snd_sc6000 *sc6000) { char answer[15]; char version[2]; - int mss_config = sc6000_irq_to_softcfg(irq[dev]) | - sc6000_dma_to_softcfg(dma[dev]); - int config = mss_config | - sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; - int old = 0; - err = sc6000_dsp_reset(vport); + err = sc6000_dsp_reset(sc6000->vport); if (err < 0) { dev_err(devptr, "sc6000_dsp_reset: failed!\n"); return err; } memset(answer, 0, sizeof(answer)); - err = sc6000_dsp_get_answer(devptr, vport, GET_DSP_COPYRIGHT, answer, 15); + err = sc6000_dsp_get_answer(devptr, sc6000->vport, GET_DSP_COPYRIGHT, + answer, 15); if (err <= 0) { dev_err(devptr, "sc6000_dsp_copyright: failed!\n"); return -ENODEV; @@ -408,54 +473,17 @@ static int sc6000_init_board(struct device *devptr, if (strncmp("SC-6000", answer, 7)) dev_warn(devptr, "Warning: non SC-6000 audio card!\n"); - if (sc6000_dsp_get_answer(devptr, vport, GET_DSP_VERSION, version, 2) < 2) { + if (sc6000_dsp_get_answer(devptr, sc6000->vport, + GET_DSP_VERSION, version, 2) < 2) { dev_err(devptr, "sc6000_dsp_version: failed!\n"); return -ENODEV; } dev_info(devptr, "Detected model: %s, DSP version %d.%d\n", answer, version[0], version[1]); - /* set configuration */ - sc6000_write(devptr, vport, COMMAND_5C); - if (sc6000_read(vport) < 0) - old = 1; - - if (!old) { - int cfg[2]; - sc6000_hw_cfg_encode(devptr, - vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev], joystick[dev]); - if (sc6000_hw_cfg_write(devptr, vport, cfg) < 0) { - dev_err(devptr, "sc6000_hw_cfg_write: failed!\n"); - return -EIO; - } - } - err = sc6000_setup_board(devptr, vport, config); - if (err < 0) { - dev_err(devptr, "sc6000_setup_board: failed!\n"); - return -ENODEV; - } - - sc6000_dsp_reset(vport); - - if (!old) { - sc6000_write(devptr, vport, COMMAND_60); - sc6000_write(devptr, vport, 0x02); - sc6000_dsp_reset(vport); - } + sc6000_detect_old_dsp(devptr, sc6000); - err = sc6000_setup_board(devptr, vport, config); - if (err < 0) { - dev_err(devptr, "sc6000_setup_board: failed!\n"); - return -ENODEV; - } - err = sc6000_init_mss(devptr, vport, config, vmss_port, mss_config); - if (err < 0) { - dev_err(devptr, "Cannot initialize Microsoft Sound System mode.\n"); - return -ENODEV; - } - - return 0; + return sc6000_program_board(devptr, sc6000); } static int snd_sc6000_mixer(struct snd_wss *chip) @@ -538,10 +566,10 @@ static int snd_sc6000_match(struct device *devptr, unsigned int dev) static void snd_sc6000_free(struct snd_card *card) { - char __iomem *vport = (char __force __iomem *)card->private_data; + struct snd_sc6000 *sc6000 = card->private_data; - if (vport) - sc6000_setup_board(card->dev, vport, 0); + if (sc6000->vport) + sc6000_setup_board(card->dev, sc6000->vport, 0); } static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) @@ -552,15 +580,17 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) int xirq = irq[dev]; int xdma = dma[dev]; struct snd_card *card; + struct snd_sc6000 *sc6000; struct snd_wss *chip; struct snd_opl3 *opl3; char __iomem *vport; char __iomem *vmss_port; err = snd_devm_card_new(devptr, index[dev], id[dev], THIS_MODULE, - 0, &card); + sizeof(*sc6000), &card); if (err < 0) return err; + sc6000 = card->private_data; if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); @@ -587,7 +617,7 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) dev_err(devptr, "I/O port cannot be iomapped.\n"); return -EBUSY; } - card->private_data = (void __force *)vport; + sc6000->vport = vport; /* to make it marked as used */ if (!devm_request_region(devptr, mss_port[dev], 4, DRV_NAME)) { @@ -600,12 +630,15 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) dev_err(devptr, "MSS port I/O cannot be iomapped.\n"); return -EBUSY; } + sc6000->vmss_port = vmss_port; dev_dbg(devptr, "Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", port[dev], xirq, xdma, mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); - err = sc6000_init_board(devptr, vport, vmss_port, dev); + sc6000_prepare_board(devptr, sc6000, dev, xirq, xdma); + + err = sc6000_init_board(devptr, sc6000); if (err < 0) return err; card->private_free = snd_sc6000_free; @@ -614,6 +647,7 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) WSS_HW_DETECT, 0, &chip); if (err < 0) return err; + sc6000->chip = chip; err = snd_wss_pcm(chip, 0); if (err < 0) { @@ -670,10 +704,47 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) return snd_card_free_on_error(devptr, __snd_sc6000_probe(devptr, dev)); } +#ifdef CONFIG_PM +static int snd_sc6000_suspend(struct device *devptr, unsigned int dev, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(devptr); + struct snd_sc6000 *sc6000 = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + sc6000->chip->suspend(sc6000->chip); + return 0; +} + +static int snd_sc6000_resume(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + struct snd_sc6000 *sc6000 = card->private_data; + int err; + + err = sc6000_dsp_reset(sc6000->vport); + if (err < 0) { + dev_err(devptr, "sc6000_dsp_reset: failed!\n"); + return err; + } + + err = sc6000_program_board(devptr, sc6000); + if (err < 0) + return err; + + sc6000->chip->resume(sc6000->chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct isa_driver snd_sc6000_driver = { .match = snd_sc6000_match, .probe = snd_sc6000_probe, - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_sc6000_suspend, + .resume = snd_sc6000_resume, +#endif .driver = { .name = DRV_NAME, }, diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index a31ca75774a6..553ceb92d298 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -131,6 +131,11 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; + unsigned long wss_base; + int irq; + int mpu_irq; + int dma1; + int dma2; int ic_type; enum card_type type; struct resource *io_res; @@ -138,6 +143,8 @@ struct soundscape { struct snd_wss *chip; unsigned char midi_vol; + bool joystick; + bool midi_enabled; struct device *dev; }; @@ -150,6 +157,21 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) } /* + * Store the resolved board settings in the per-card state so that + * the same configuration can be replayed later if necessary. + */ +static void sscape_store_settings(struct soundscape *sscape, int dev) +{ + sscape->io_base = port[dev]; + sscape->wss_base = wss_port[dev]; + sscape->irq = irq[dev]; + sscape->mpu_irq = mpu_irq[dev]; + sscape->dma1 = dma[dev]; + sscape->dma2 = dma2[dev]; + sscape->joystick = joystick[dev]; +} + +/* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to * contiguous pages of physical memory. @@ -263,34 +285,36 @@ static int host_read_ctrl_unsafe(unsigned io_base, unsigned timeout) /* * Write to the SoundScape's host-mode control registers, but - * leave any locking issues to the caller ... + * leave any locking issues to the caller. Returns true if + * the write succeeded. */ -static inline int host_write_unsafe(unsigned io_base, unsigned char data) +static inline bool host_write_unsafe(unsigned int io_base, unsigned char data) { if ((inb(HOST_CTRL_IO(io_base)) & TX_READY) != 0) { outb(data, HOST_DATA_IO(io_base)); - return 1; + return true; } - return 0; + return false; } /* * Write to the SoundScape's host-mode control registers, performing * a limited amount of busy-waiting if the register isn't ready. - * Also leaves all locking-issues to the caller ... + * Also leaves all locking-issues to the caller. Returns true if + * the write succeeded before timing out. */ -static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) +static bool host_write_ctrl_unsafe(unsigned int io_base, unsigned char data, + unsigned int timeout) { - int err; + bool written; - while (!(err = host_write_unsafe(io_base, data)) && (timeout != 0)) { + while (!(written = host_write_unsafe(io_base, data)) && timeout != 0) { udelay(100); --timeout; } /* while */ - return err; + return written; } @@ -561,6 +585,30 @@ static int sscape_upload_microcode(struct snd_card *card, int version) } /* + * Restore the SoundScape's MIDI control state after the firmware + * upload has made the host interface available again. + */ +static int sscape_restore_midi_state(struct soundscape *sscape) +{ + bool success; + + guard(spinlock_irqsave)(&sscape->lock); + set_host_mode_unsafe(sscape->io_base); + + success = host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100) && + host_write_ctrl_unsafe(sscape->io_base, sscape->midi_vol, 100) && + host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100) && + host_write_ctrl_unsafe(sscape->io_base, sscape->midi_vol, 100) && + host_write_ctrl_unsafe(sscape->io_base, CMD_SET_EXTMIDI, 100) && + host_write_ctrl_unsafe(sscape->io_base, 0, 100) && + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + + return success ? 0 : -EIO; +} + +/* * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, @@ -661,6 +709,59 @@ static unsigned get_irq_config(int sscape_type, int irq) } /* + * Program the SoundScape's board-specific routing and enable the + * codec path using the resolved IRQ, DMA and joystick settings. + */ +static int sscape_configure_board(struct soundscape *sscape) +{ + unsigned int dma_cfg; + unsigned int irq_cfg; + unsigned int mpu_irq_cfg; + int val; + + irq_cfg = get_irq_config(sscape->type, sscape->irq); + if (irq_cfg == INVALID_IRQ) + return -ENXIO; + + mpu_irq_cfg = get_irq_config(sscape->type, sscape->mpu_irq); + if (mpu_irq_cfg == INVALID_IRQ) + return -ENXIO; + + scoped_guard(spinlock_irqsave, &sscape->lock) { + if (sscape->ic_type == IC_OPUS) + activate_ad1845_unsafe(sscape->io_base); + + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); + sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); + + /* + * Enable and configure the DMA channels ... + */ + sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); + sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); + sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); + + mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xf7; + if (sscape->joystick) + val |= 0x08; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0xd0); + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, + 0xf0 | mpu_irq_cfg); + sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, + 0x09 | DMA_8BIT | + (sscape->dma1 << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); + } + + return 0; +} + +/* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ @@ -890,37 +991,33 @@ _error: /* * Create an ALSA soundcard entry for the SoundScape, using - * the given list of port, IRQ and DMA resources. + * the resolved port, IRQ and DMA resources. */ -static int create_sscape(int dev, struct snd_card *card) +static int create_sscape(struct snd_card *card) { struct soundscape *sscape = get_card_soundscape(card); - unsigned dma_cfg; - unsigned irq_cfg; - unsigned mpu_irq_cfg; struct resource *io_res; struct resource *wss_res; int err; - int val; const char *name; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = devm_request_region(card->dev, port[dev], 8, "SoundScape"); + io_res = devm_request_region(card->dev, sscape->io_base, 8, "SoundScape"); if (!io_res) { dev_err(card->dev, - "sscape: can't grab port 0x%lx\n", port[dev]); + "sscape: can't grab port 0x%x\n", sscape->io_base); return -EBUSY; } wss_res = NULL; if (sscape->type == SSCAPE_VIVO) { - wss_res = devm_request_region(card->dev, wss_port[dev], 4, + wss_res = devm_request_region(card->dev, sscape->wss_base, 4, "SoundScape"); if (!wss_res) { dev_err(card->dev, "sscape: can't grab port 0x%lx\n", - wss_port[dev]); + sscape->wss_base); return -EBUSY; } } @@ -928,18 +1025,17 @@ static int create_sscape(int dev, struct snd_card *card) /* * Grab one DMA channel ... */ - err = snd_devm_request_dma(card->dev, dma[dev], "SoundScape"); + err = snd_devm_request_dma(card->dev, sscape->dma1, "SoundScape"); if (err < 0) { - dev_err(card->dev, "sscape: can't grab DMA %d\n", dma[dev]); + dev_err(card->dev, "sscape: can't grab DMA %d\n", sscape->dma1); return err; } spin_lock_init(&sscape->lock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = port[dev]; - if (!detect_sscape(sscape, wss_port[dev])) { + if (!detect_sscape(sscape, sscape->wss_base)) { dev_err(card->dev, "sscape: hardware not detected at 0x%x\n", sscape->io_base); return -ENODEV; @@ -964,66 +1060,28 @@ static int create_sscape(int dev, struct snd_card *card) } dev_info(card->dev, "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", - name, sscape->io_base, irq[dev], dma[dev]); - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(sscape->type, irq[dev]); - if (irq_cfg == INVALID_IRQ) { - dev_err(card->dev, "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - dev_err(card->dev, "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + name, sscape->io_base, sscape->irq, sscape->dma1); /* * Tell the on-board devices where their resources are (I think - * I can't be sure without a datasheet ... So many magic values!) */ - scoped_guard(spinlock_irqsave, &sscape->lock) { - - sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); - sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); - - /* - * Enable and configure the DMA channels ... - */ - sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); - sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); - sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - - mpu_irq_cfg |= mpu_irq_cfg << 2; - val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; - if (joystick[dev]) - val |= 8; - sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); - sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); - sscape_write_unsafe(sscape->io_base, - GA_CDCFG_REG, 0x09 | DMA_8BIT - | (dma[dev] << 4) | (irq_cfg << 1)); - /* - * Enable the master IRQ ... - */ - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); - + err = sscape_configure_board(sscape); + if (err < 0) { + dev_err(card->dev, "sscape: Invalid IRQ configuration\n"); + return err; } /* * We have now enabled the codec chip, and so we should * detect the AD1845 device ... */ - err = create_ad1845(card, wss_port[dev], irq[dev], - dma[dev], dma2[dev]); + err = create_ad1845(card, sscape->wss_base, sscape->irq, + sscape->dma1, sscape->dma2); if (err < 0) { dev_err(card->dev, "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + sscape->wss_base, sscape->irq); return err; } strscpy(card->driver, "SoundScape"); @@ -1040,41 +1098,99 @@ static int create_sscape(int dev, struct snd_card *card) err = sscape_upload_microcode(card, err); if (err == 0) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], - mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, sscape->io_base, + sscape->mpu_irq); if (err < 0) { dev_err(card->dev, "sscape: Failed to create MPU-401 device at 0x%lx\n", - port[dev]); + (unsigned long)sscape->io_base); return err; } - /* - * Initialize mixer - */ - guard(spinlock_irqsave)(&sscape->lock); sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, - CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, - sscape->midi_vol, 100); - host_write_ctrl_unsafe(sscape->io_base, - CMD_XXX_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, - sscape->midi_vol, 100); - host_write_ctrl_unsafe(sscape->io_base, - CMD_SET_EXTMIDI, 100); - host_write_ctrl_unsafe(sscape->io_base, - 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); - - set_midi_mode_unsafe(sscape->io_base); + sscape->midi_enabled = true; + err = sscape_restore_midi_state(sscape); + if (err < 0) + dev_warn(card->dev, + "sscape: MIDI init incomplete: %d\n", + err); } } return 0; } +#ifdef CONFIG_PM +/* + * Reload the MIDI firmware and restore the saved MIDI state for + * boards whose MPU-401 side was enabled during probe. + */ +static int sscape_resume_midi(struct snd_card *card) +{ + struct soundscape *sscape = get_card_soundscape(card); + int err, version; + + if (!sscape->midi_enabled) + return 0; + + version = sscape_upload_bootblock(card); + if (version < 0) + return version; + + err = sscape_upload_microcode(card, version); + if (err < 0) + return err; + + outb(0, sscape->io_base); + + return sscape_restore_midi_state(sscape); +} + +/* + * Save the WSS codec state before the SoundScape is suspended. + */ +static int snd_sscape_suspend_card(struct snd_card *card) +{ + struct soundscape *sscape = get_card_soundscape(card); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + sscape->chip->suspend(sscape->chip); + return 0; +} + +/* + * Restore the board-specific state before resuming the WSS codec. + */ +static int snd_sscape_resume_card(struct snd_card *card) +{ + struct soundscape *sscape = get_card_soundscape(card); + int err; + + err = sscape_configure_board(sscape); + if (err < 0) + return err; + + err = sscape_resume_midi(card); + if (err < 0) + dev_warn(card->dev, "sscape: MIDI restore failed: %d\n", err); + + sscape->chip->resume(sscape->chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static int snd_sscape_suspend(struct device *dev, unsigned int n, + pm_message_t state) +{ + return snd_sscape_suspend_card(dev_get_drvdata(dev)); +} + +static int snd_sscape_resume(struct device *dev, unsigned int n) +{ + return snd_sscape_resume_card(dev_get_drvdata(dev)); +} +#endif + static int snd_sscape_match(struct device *pdev, unsigned int i) { @@ -1111,8 +1227,9 @@ static int snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + sscape_store_settings(sscape, dev); - ret = create_sscape(dev, card); + ret = create_sscape(card); if (ret < 0) return ret; @@ -1130,7 +1247,10 @@ static int snd_sscape_probe(struct device *pdev, unsigned int dev) static struct isa_driver snd_sscape_driver = { .match = snd_sscape_match, .probe = snd_sscape_probe, - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_sscape_suspend, + .resume = snd_sscape_resume, +#endif .driver = { .name = DEV_NAME }, @@ -1211,8 +1331,9 @@ static int sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + sscape_store_settings(sscape, idx); - ret = create_sscape(idx, card); + ret = create_sscape(card); if (ret < 0) return ret; @@ -1227,11 +1348,27 @@ static int sscape_pnp_detect(struct pnp_card_link *pcard, return 0; } +#ifdef CONFIG_PM +static int sscape_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state) +{ + return snd_sscape_suspend_card(pnp_get_card_drvdata(pcard)); +} + +static int sscape_pnp_resume(struct pnp_card_link *pcard) +{ + return snd_sscape_resume_card(pnp_get_card_drvdata(pcard)); +} +#endif + static struct pnp_card_driver sscape_pnpc_driver = { .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, .name = "sscape", .id_table = sscape_pnpids, .probe = sscape_pnp_detect, +#ifdef CONFIG_PM + .suspend = sscape_pnp_suspend, + .resume = sscape_pnp_resume, +#endif }; #endif /* CONFIG_PNP */ diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 3a64d0562803..b1c7ed7f1604 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1362,6 +1362,7 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control, struct hpi_control *hpi_ctl, char *name) { + int len; char *dir; memset(snd_control, 0, sizeof(*snd_control)); snd_control->name = hpi_ctl->name; @@ -1384,23 +1385,30 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control, dir = "Playback "; /* PCM Playback source, or output node */ if (hpi_ctl->src_node_type && hpi_ctl->dst_node_type) - sprintf(hpi_ctl->name, "%s %d %s %d %s%s", - asihpi_src_names[hpi_ctl->src_node_type], - hpi_ctl->src_node_index, - asihpi_dst_names[hpi_ctl->dst_node_type], - hpi_ctl->dst_node_index, - dir, name); + len = snprintf(hpi_ctl->name, sizeof(hpi_ctl->name), + "%s %d %s %d %s%s", + asihpi_src_names[hpi_ctl->src_node_type], + hpi_ctl->src_node_index, + asihpi_dst_names[hpi_ctl->dst_node_type], + hpi_ctl->dst_node_index, + dir, name); else if (hpi_ctl->dst_node_type) { - sprintf(hpi_ctl->name, "%s %d %s%s", - asihpi_dst_names[hpi_ctl->dst_node_type], - hpi_ctl->dst_node_index, - dir, name); + len = snprintf(hpi_ctl->name, sizeof(hpi_ctl->name), + "%s %d %s%s", + asihpi_dst_names[hpi_ctl->dst_node_type], + hpi_ctl->dst_node_index, + dir, name); } else { - sprintf(hpi_ctl->name, "%s %d %s%s", - asihpi_src_names[hpi_ctl->src_node_type], - hpi_ctl->src_node_index, - dir, name); + len = snprintf(hpi_ctl->name, sizeof(hpi_ctl->name), + "%s %d %s%s", + asihpi_src_names[hpi_ctl->src_node_type], + hpi_ctl->src_node_index, + dir, name); } + + if (len >= sizeof(hpi_ctl->name)) + pr_err("asihpi: truncated control name: %s\n", + hpi_ctl->name); } /*------------------------------------------------------------ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index da2667cb2489..02fe09330939 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -794,7 +794,8 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, struct src *src; int err; int n_amixer = apcm->substream->runtime->channels, i; - unsigned int pitch, rsr = atc->pll_rate; + unsigned int pitch; + unsigned int rsr = atc->pll_rate ? atc->pll_rate : atc->rsr; /* first release old resources */ atc_pcm_release_resources(atc, apcm); @@ -983,6 +984,11 @@ static int atc_select_mic_in(struct ct_atc *atc) return 0; } +static inline enum DAIOTYP atc_spdif_in_type(struct ct_atc *atc) +{ + return (atc->model == CTSB073X) ? SPDIFI_BAY : SPDIFIO; +} + static struct capabilities atc_capabilities(struct ct_atc *atc) { struct hw *hw = atc->hw; @@ -1121,7 +1127,7 @@ static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state) static int atc_spdif_in_unmute(struct ct_atc *atc, unsigned char state) { - return atc_daio_unmute(atc, state, SPDIFIO); + return atc_daio_unmute(atc, state, atc_spdif_in_type(atc)); } static int atc_spdif_out_get_status(struct ct_atc *atc, unsigned int *status) @@ -1404,9 +1410,11 @@ static int atc_get_resources(struct ct_atc *atc) struct sum_desc sum_dsc = {0}; struct sum_mgr *sum_mgr; struct capabilities cap; + int atc_srcs_limit; int err, i; cap = atc->capabilities(atc); + atc_srcs_limit = cap.dedicated_mic ? NUM_ATC_SRCS : 4; atc->daios = kcalloc(NUM_DAIOTYP, sizeof(void *), GFP_KERNEL); if (!atc->daios) @@ -1427,14 +1435,12 @@ static int atc_get_resources(struct ct_atc *atc) daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; da_desc.msr = atc->msr; for (i = 0; i < NUM_DAIOTYP; i++) { - if (((i == MIC) && !cap.dedicated_mic) || - ((i == RCA) && !cap.dedicated_rca) || - i == SPDIFI1) + if (((i == SPDIFIO) && (atc->model == CTSB073X)) || + ((i == SPDIFI_BAY) && (atc->model != CTSB073X)) || + ((i == MIC) && !cap.dedicated_mic) || + ((i == RCA) && !cap.dedicated_rca)) continue; - if (atc->model == CTSB073X && i == SPDIFIO) - da_desc.type = SPDIFI1; - else - da_desc.type = i; + da_desc.type = i; da_desc.output = (i < LINEIM) || (i == RCA); err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); @@ -1450,9 +1456,7 @@ static int atc_get_resources(struct ct_atc *atc) src_dsc.multi = 1; src_dsc.msr = atc->msr; src_dsc.mode = ARCRW; - for (i = 0; i < NUM_ATC_SRCS; i++) { - if (((i > 3) && !cap.dedicated_mic)) - continue; + for (i = 0; i < atc_srcs_limit; i++) { err = src_mgr->get_src(src_mgr, &src_dsc, (struct src **)&atc->srcs[i]); if (err) @@ -1461,9 +1465,7 @@ static int atc_get_resources(struct ct_atc *atc) srcimp_mgr = atc->rsc_mgrs[SRCIMP]; srcimp_dsc.msr = 8; - for (i = 0; i < NUM_ATC_SRCS; i++) { - if (((i > 3) && !cap.dedicated_mic)) - continue; + for (i = 0; i < atc_srcs_limit; i++) { err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, (struct srcimp **)&atc->srcimps[i]); if (err) @@ -1569,7 +1571,7 @@ static void atc_connect_resources(struct ct_atc *atc) mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc); } - dai = container_of(atc->daios[SPDIFIO], struct dai, daio); + dai = container_of(atc->daios[atc_spdif_in_type(atc)], struct dai, daio); atc_connect_dai(atc->rsc_mgrs[SRC], dai, (struct src **)&atc->srcs[0], (struct srcimp **)&atc->srcimps[0]); diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 4dbb1dd7af32..69aacd06716c 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -35,7 +35,7 @@ static const struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = { [LINEIM] = {.left = 0x1b5, .right = 0x1bd}, [SPDIFOO] = {.left = 0x20, .right = 0x21}, [SPDIFIO] = {.left = 0x15, .right = 0x1d}, - [SPDIFI1] = {.left = 0x95, .right = 0x9d}, + [SPDIFI_BAY] = {.left = 0x95, .right = 0x9d}, }; static const struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { @@ -106,7 +106,7 @@ static int daio_device_index(enum DAIOTYP type, struct hw *hw) switch (type) { case SPDIFOO: return 0; case SPDIFIO: return 0; - case SPDIFI1: return 1; + case SPDIFI_BAY: return 1; case LINEO1: return 4; case LINEO2: return 7; case LINEO3: return 5; @@ -120,7 +120,6 @@ static int daio_device_index(enum DAIOTYP type, struct hw *hw) switch (type) { case SPDIFOO: return 0; case SPDIFIO: return 0; - case SPDIFI1: return 1; case LINEO1: return 4; case LINEO2: return 7; case LINEO3: return 5; diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h index ff77d55539a5..c9f6207fe92f 100644 --- a/sound/pci/ctxfi/ctdaio.h +++ b/sound/pci/ctxfi/ctdaio.h @@ -32,7 +32,7 @@ enum DAIOTYP { SPDIFIO, /* S/PDIF In (Flexijack/Optical) on the card */ MIC, /* Dedicated mic on Titanium HD */ RCA, /* Dedicated RCA on SE-300PCIE */ - SPDIFI1, /* S/PDIF In on internal Drive Bay */ + SPDIFI_BAY, /* S/PDIF In on internal drive bay */ NUM_DAIOTYP }; diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h index da54cbcdb0be..43a0065b40c3 100644 --- a/sound/pci/ctxfi/ctvmem.h +++ b/sound/pci/ctxfi/ctvmem.h @@ -15,7 +15,7 @@ #ifndef CTVMEM_H #define CTVMEM_H -#define CT_PTP_NUM 4 /* num of device page table pages */ +#define CT_PTP_NUM 1 /* num of device page table pages */ #include <linux/mutex.h> #include <linux/list.h> diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index 5ff78814e687..874f6cd503ca 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -53,11 +53,6 @@ static void usb6fire_chip_abort(struct sfire_chip *chip) usb6fire_comm_abort(chip); if (chip->control) usb6fire_control_abort(chip); - if (chip->card) { - snd_card_disconnect(chip->card); - snd_card_free_when_closed(chip->card); - chip->card = NULL; - } } } @@ -168,6 +163,7 @@ destroy_chip: static void usb6fire_chip_disconnect(struct usb_interface *intf) { struct sfire_chip *chip; + struct snd_card *card; chip = usb_get_intfdata(intf); if (chip) { /* if !chip, fw upload has been performed */ @@ -178,8 +174,19 @@ static void usb6fire_chip_disconnect(struct usb_interface *intf) chips[chip->regidx] = NULL; } + /* + * Save card pointer before teardown. + * snd_card_free_when_closed() may free card (and + * the embedded chip) immediately, so it must be + * called last and chip must not be accessed after. + */ + card = chip->card; chip->shutdown = true; + if (card) + snd_card_disconnect(card); usb6fire_chip_abort(chip); + if (card) + snd_card_free_when_closed(card); } } } diff --git a/sound/usb/card.c b/sound/usb/card.c index 270dad84d825..f42d72cd0378 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -631,9 +631,9 @@ static void usb_audio_make_shortname(struct usb_device *dev, } /* retrieve the device string as shortname */ - if (!dev->descriptor.iProduct || - usb_string(dev, dev->descriptor.iProduct, - card->shortname, sizeof(card->shortname)) <= 0) { + if (dev->product && *dev->product) { + strscpy(card->shortname, dev->product); + } else { /* no name available from anywhere, so use ID */ scnprintf(card->shortname, sizeof(card->shortname), "USB Device %#04x:%#04x", @@ -668,15 +668,11 @@ static void usb_audio_make_longname(struct usb_device *dev, else if (quirk && quirk->vendor_name) s = quirk->vendor_name; *card->longname = 0; - if (s && *s) { - strscpy(card->longname, s, sizeof(card->longname)); - } else { - /* retrieve the vendor and device strings as longname */ - if (dev->descriptor.iManufacturer) - usb_string(dev, dev->descriptor.iManufacturer, - card->longname, sizeof(card->longname)); - /* we don't really care if there isn't any vendor string */ - } + if (s && *s) + strscpy(card->longname, s); + else if (dev->manufacturer && *dev->manufacturer) + strscpy(card->longname, dev->manufacturer); + if (*card->longname) { strim(card->longname); if (*card->longname) @@ -870,19 +866,25 @@ static void find_last_interface(struct snd_usb_audio *chip) /* look for the corresponding quirk */ static const struct snd_usb_audio_quirk * -get_alias_quirk(struct usb_device *dev, unsigned int id) +get_alias_quirk(struct usb_interface *intf, unsigned int id) { const struct usb_device_id *p; + struct usb_device_id match_id; for (p = usb_audio_ids; p->match_flags; p++) { - /* FIXME: this checks only vendor:product pair in the list */ - if ((p->match_flags & USB_DEVICE_ID_MATCH_DEVICE) == - USB_DEVICE_ID_MATCH_DEVICE && - p->idVendor == USB_ID_VENDOR(id) && - p->idProduct == USB_ID_PRODUCT(id)) - return (const struct snd_usb_audio_quirk *)p->driver_info; - } + if ((p->match_flags & USB_DEVICE_ID_MATCH_DEVICE) != + USB_DEVICE_ID_MATCH_DEVICE) + continue; + if (p->idVendor != USB_ID_VENDOR(id) || + p->idProduct != USB_ID_PRODUCT(id)) + continue; + match_id = *p; + match_id.match_flags &= ~USB_DEVICE_ID_MATCH_DEVICE; + if (!match_id.match_flags || usb_match_one_id(intf, &match_id)) + return (const struct snd_usb_audio_quirk *) + p->driver_info; + } return NULL; } @@ -931,7 +933,7 @@ static int usb_audio_probe(struct usb_interface *intf, id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); if (get_alias_id(dev, &id)) - quirk = get_alias_quirk(dev, id); + quirk = get_alias_quirk(intf, id); if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) return -ENXIO; if (quirk && quirk->ifnum == QUIRK_NODEV_INTERFACE) diff --git a/sound/usb/format.c b/sound/usb/format.c index 1207c507882a..030b4307927a 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -455,6 +455,10 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x194f, 0x010d) && !s1810c_valid_sample_rate(fp, rate)) goto skip_rate; + /* Filter out invalid rates on Presonus Studio 1824 */ + if (chip->usb_id == USB_ID(0x194f, 0x0107) && + !s1810c_valid_sample_rate(fp, rate)) + goto skip_rate; /* Filter out invalid rates on Focusrite devices */ if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a8bddc90c0ed..0a5b8941ebda 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -699,15 +699,18 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port *port, static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, struct urb *urb) { - int p; + int port0 = ep->current_port; + int i; + + for (i = 0; i < 0x10; ++i) { + int portnum = (port0 + i) & 15; + struct usbmidi_out_port *port = &ep->ports[portnum]; - /* FIXME: lower-numbered ports can starve higher-numbered ports */ - for (p = 0; p < 0x10; ++p) { - struct usbmidi_out_port *port = &ep->ports[p]; if (!port->active) continue; while (urb->transfer_buffer_length + 3 < ep->max_transfer) { uint8_t b; + if (snd_rawmidi_transmit(port->substream, &b, 1) != 1) { port->active = 0; break; @@ -715,6 +718,7 @@ static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, snd_usbmidi_transmit_byte(port, b, urb); } } + ep->current_port = (port0 + 1) & 15; } static const struct usb_protocol_ops snd_usbmidi_standard_ops = { diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index ef602e81576d..3546ba926cb3 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -1057,10 +1057,8 @@ static void set_fallback_rawmidi_names(struct snd_usb_midi2_interface *umidi) strscpy(ump->core.name, ump->info.name, sizeof(ump->core.name)); /* use serial number string as unique UMP product id */ - if (!*ump->info.product_id && dev->descriptor.iSerialNumber) - usb_string(dev, dev->descriptor.iSerialNumber, - ump->info.product_id, - sizeof(ump->info.product_id)); + if (!*ump->info.product_id && dev->serial && *dev->serial) + strscpy(ump->info.product_id, dev->serial); } } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ac8c71ba9483..d4ef45bf53d7 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1204,6 +1204,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->min = -11264; /* Mute under it */ } break; + case USB_ID(0x31b2, 0x0111): /* MOONDROP JU Jiu */ + if (!strcmp(kctl->id.name, "PCM Playback Volume")) { + usb_audio_info(chip, + "set volume quirk for MOONDROP JU Jiu\n"); + cval->min = -10880; /* Mute under it */ + } + break; } } @@ -1226,12 +1233,79 @@ static void init_cur_mix_raw(struct usb_mixer_elem_info *cval, int ch, int idx) } /* + * Additional checks for sticky mixers + * + * Some devices' volume control mixers are sticky, which accept SET_CUR but + * do absolutely nothing. + * + * Prevent sticky mixers from being registered, otherwise they confuses + * userspace and results in ineffective volume control. + */ +static int check_sticky_volume_control(struct usb_mixer_elem_info *cval, + int channel, int saved) +{ + int sticky_test_values[] = { cval->min, cval->max }; + int test, check, i; + + for (i = 0; i < ARRAY_SIZE(sticky_test_values); i++) { + test = sticky_test_values[i]; + if (test == saved) + continue; + + /* Assume non-sticky on failure. */ + if (snd_usb_set_cur_mix_value(cval, channel, 0, test) || + get_cur_mix_raw(cval, channel, &check) || + check != saved) /* SET_CUR effective, non-sticky. */ + return 0; + } + + usb_audio_err(cval->head.mixer->chip, + "%d:%d: sticky mixer values (%d/%d/%d => %d), disabling\n", + cval->head.id, mixer_ctrl_intf(cval->head.mixer), + cval->min, cval->max, cval->res, saved); + + return -ENODEV; +} + +/* + * Additional checks for the proper resolution + * + * Some devices report smaller resolutions than actually reacting. + * They don't return errors but simply clip to the lower aligned value. + */ +static void check_volume_control_res(struct usb_mixer_elem_info *cval, + int channel, int saved) +{ + int last_valid_res = cval->res; + int test, check; + + for (;;) { + test = saved; + if (test < cval->max) + test += cval->res; + else + test -= cval->res; + + if (test < cval->min || test > cval->max || + snd_usb_set_cur_mix_value(cval, channel, 0, test) || + get_cur_mix_raw(cval, channel, &check)) { + cval->res = last_valid_res; + break; + } + if (test == check) + break; + + cval->res *= 2; + } +} + +/* * retrieve the minimum and maximum values for the specified control */ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, int default_min, struct snd_kcontrol *kctl) { - int i, idx; + int i, idx, ret; /* for failsafe */ cval->min = default_min; @@ -1257,7 +1331,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->head.id, mixer_ctrl_intf(cval->head.mixer), cval->control, cval->head.id); - return -EINVAL; + return -EAGAIN; } if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, @@ -1280,37 +1354,25 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, if (cval->res == 0) cval->res = 1; - /* Additional checks for the proper resolution - * - * Some devices report smaller resolutions than actually - * reacting. They don't return errors but simply clip - * to the lower aligned value. - */ - if (cval->min + cval->res < cval->max) { - int last_valid_res = cval->res; - int saved, test, check; + if (cval->min < cval->max) { + int saved; + if (get_cur_mix_raw(cval, minchn, &saved) < 0) - goto no_res_check; - for (;;) { - test = saved; - if (test < cval->max) - test += cval->res; - else - test -= cval->res; - if (test < cval->min || test > cval->max || - snd_usb_set_cur_mix_value(cval, minchn, 0, test) || - get_cur_mix_raw(cval, minchn, &check)) { - cval->res = last_valid_res; - break; - } - if (test == check) - break; - cval->res *= 2; + goto no_checks; + + ret = check_sticky_volume_control(cval, minchn, saved); + if (ret < 0) { + snd_usb_set_cur_mix_value(cval, minchn, 0, saved); + return ret; } + + if (cval->min + cval->res < cval->max) + check_volume_control_res(cval, minchn, saved); + snd_usb_set_cur_mix_value(cval, minchn, 0, saved); } -no_res_check: +no_checks: cval->initialized = 1; } @@ -1381,6 +1443,7 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *cval = snd_kcontrol_chip(kcontrol); + int ret; if (cval->val_type == USB_MIXER_BOOLEAN || cval->val_type == USB_MIXER_INV_BOOLEAN) @@ -1391,8 +1454,9 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, if (cval->val_type != USB_MIXER_BOOLEAN && cval->val_type != USB_MIXER_INV_BOOLEAN) { if (!cval->initialized) { - get_min_max_with_quirks(cval, 0, kcontrol); - if (cval->initialized && cval->dBmin >= cval->dBmax) { + ret = get_min_max_with_quirks(cval, 0, kcontrol); + if ((ret >= 0 || ret == -EAGAIN) && + cval->initialized && cval->dBmin >= cval->dBmax) { kcontrol->vd[0].access &= ~(SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); @@ -1660,9 +1724,72 @@ static const struct usb_feature_control_info *get_feature_control_info(int contr return NULL; } +static bool check_insane_volume_range(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, + struct usb_mixer_elem_info *cval) +{ + int range, steps, threshold; + + /* + * If a device quirk has overrode our TLV callback, no warning should + * be generated since our checks are only meaningful for dB volume. + */ + if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) || + kctl->tlv.c != snd_usb_mixer_vol_tlv) + return false; + + /* + * Meaningless volume control capability (<1dB). This should cover + * devices mapping their volume to val = 0/100/1, which are very likely + * to be quirky. + */ + range = cval->max - cval->min; + if (range < 256) { + usb_audio_warn(mixer->chip, + "Warning! Unlikely small volume range (=%u), linear volume or custom curve?", + range); + return true; + } + + steps = range / cval->res; + + /* + * There are definitely devices with ~20,000 ranges (e.g., HyperX Cloud + * III with val = -18944/0/1), so we use some heuristics here: + * + * min < 0 < max: Attenuator + amplifier? Likely to be sane + * + * min < 0 = max: DSP? Voltage attenuator with FW conversion to dB? + * Likely to be sane + * + * min < max < 0: Measured values? Neutral + * + * min = 0 < max: Oversimplified FW conversion? Linear volume? Likely to + * be quirky (e.g., MV-SILICON) + * + * 0 < min < max: Amplifier with fixed gains? Likely to be quirky + * (e.g., Logitech webcam) + */ + if (cval->min < 0 && 0 <= cval->max) + threshold = 24576; /* 65535 * (3 / 8) */ + else if (cval->min < cval->max && cval->max < 0) + threshold = 1024; + else + threshold = 384; + + if (steps > threshold) { + usb_audio_warn(mixer->chip, + "Warning! Unlikely big volume step count (=%u), linear volume or wrong cval->res?", + steps); + return true; + } + + return false; +} + static void __build_feature_ctl(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, - unsigned int ctl_mask, int control, + u64 ctl_mask, int control, struct usb_audio_term *iterm, struct usb_audio_term *oterm, int unitid, int nameid, int readonly_mask) @@ -1673,7 +1800,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; const struct usbmix_name_map *map; - unsigned int range; + int ret; if (control == UAC_FU_GRAPHIC_EQUALIZER) { /* FIXME: not supported yet */ @@ -1707,7 +1834,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, cval->master_readonly = readonly_mask; } else { int i, c = 0; - for (i = 0; i < 16; i++) + for (i = 0; i < MAX_CHANNELS; i++) if (ctl_mask & BIT(i)) c++; cval->channels = c; @@ -1787,10 +1914,10 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, } /* get min/max values */ - get_min_max_with_quirks(cval, 0, kctl); + ret = get_min_max_with_quirks(cval, 0, kctl); /* skip a bogus volume range */ - if (cval->max <= cval->min) { + if ((ret < 0 && ret != -EAGAIN) || cval->max <= cval->min) { usb_audio_dbg(mixer->chip, "[%d] FU [%s] skipped due to invalid volume\n", cval->head.id, kctl->id.name); @@ -1811,29 +1938,21 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, snd_usb_mixer_fu_apply_quirk(mixer, cval, unitid, kctl); - range = (cval->max - cval->min) / cval->res; - /* - * There are definitely devices with a range of ~20,000, so let's be - * conservative and allow for a bit more. - */ - if (range > 65535) { - usb_audio_warn(mixer->chip, - "Warning! Unlikely big volume range (=%u), cval->res is probably wrong.", - range); - usb_audio_warn(mixer->chip, - "[%d] FU [%s] ch = %d, val = %d/%d/%d", + if (check_insane_volume_range(mixer, kctl, cval)) { + usb_audio_warn(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", cval->head.id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res); + } else { + usb_audio_dbg(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", + cval->head.id, kctl->id.name, cval->channels, + cval->min, cval->max, cval->res); } - usb_audio_dbg(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", - cval->head.id, kctl->id.name, cval->channels, - cval->min, cval->max, cval->res); snd_usb_mixer_add_control(&cval->head, kctl); } static void build_feature_ctl(struct mixer_build *state, void *raw_desc, - unsigned int ctl_mask, int control, + u64 ctl_mask, int control, struct usb_audio_term *iterm, int unitid, int readonly_mask) { @@ -1845,7 +1964,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } static void build_feature_ctl_badd(struct usb_mixer_interface *mixer, - unsigned int ctl_mask, int control, int unitid, + u64 ctl_mask, int control, int unitid, const struct usbmix_name_map *badd_map) { __build_feature_ctl(mixer, badd_map, ctl_mask, control, @@ -2021,7 +2140,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, bmaControls = ftr->bmaControls; } - if (channels > 32) { + if (channels > MAX_CHANNELS) { usb_audio_info(state->chip, "usbmixer: too many channels (%d) in unit %d\n", channels, unitid); @@ -2059,7 +2178,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, if (state->mixer->protocol == UAC_VERSION_1) { /* check all control types */ for (i = 0; i < 10; i++) { - unsigned int ch_bits = 0; + u64 ch_bits = 0; int control = audio_feature_info[i].control; for (j = 0; j < channels; j++) { @@ -2085,7 +2204,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, } } else { /* UAC_VERSION_2/3 */ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { - unsigned int ch_bits = 0; + u64 ch_bits = 0; unsigned int ch_read_only = 0; int control = audio_feature_info[i].control; @@ -2172,6 +2291,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned int i, len; struct snd_kcontrol *kctl; const struct usbmix_name_map *map; + int ret; map = find_map(state->map, unitid, 0); if (check_ignored_ctl(map)) @@ -2194,7 +2314,11 @@ static void build_mixer_unit_ctl(struct mixer_build *state, } /* get min/max values */ - get_min_max(cval, 0); + ret = get_min_max(cval, 0); + if (ret < 0 && ret != -EAGAIN) { + usb_mixer_elem_info_free(cval); + return; + } kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); if (!kctl) { @@ -2566,7 +2690,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } - get_min_max(cval, valinfo->min_value); + err = get_min_max(cval, valinfo->min_value); break; } case USB_XU_CLOCK_RATE: @@ -2578,11 +2702,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, cval->max = 5; cval->res = 1; cval->initialized = 1; + err = 0; break; default: - get_min_max(cval, valinfo->min_value); + err = get_min_max(cval, valinfo->min_value); break; } + if (err < 0 && err != -EAGAIN) { + usb_mixer_elem_info_free(cval); + return err; + } err = get_cur_ctl_value(cval, cval->control << 8, &val); if (err < 0) { @@ -3398,7 +3527,7 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, [USB_MIXER_U32] = "U32", [USB_MIXER_BESPOKEN] = "BESPOKEN", }; - snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " + snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%llx, " "channels=%i, type=\"%s\"\n", cval->head.id, cval->control, cval->cmask, cval->channels, val_types[cval->val_type]); diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 167fbfcf01ac..afbb3dd9f177 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -44,7 +44,7 @@ struct usb_mixer_interface { void (*private_suspend)(struct usb_mixer_interface *mixer); }; -#define MAX_CHANNELS 16 /* max logical channels */ +#define MAX_CHANNELS 64 /* max logical channels */ enum { USB_MIXER_BOOLEAN, @@ -81,7 +81,7 @@ struct usb_mixer_elem_list { struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ - unsigned int cmask; /* channel mask bitmap: 0 = master */ + u64 cmask; /* channel mask bitmap: 0 = master */ unsigned int idx_off; /* Control index offset */ unsigned int ch_readonly; unsigned int master_readonly; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 11e205da7964..a01510a855c2 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -4477,6 +4477,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x194f, 0x010d): /* Presonus Studio 1824c */ err = snd_sc1810_init_mixer(mixer); break; + case USB_ID(0x194f, 0x0107): /* Presonus Studio 1824 */ + err = snd_sc1810_init_mixer(mixer); + break; case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; @@ -4588,6 +4591,24 @@ static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, } } +static void snd_usb_mv_silicon_quirks(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + if (cval->min == 0 && cval->max == 4096 && cval->res == 1) { + /* The final effects will be printed later. */ + usb_audio_info(mixer->chip, "applying MV-SILICON quirks (0/4096/1 variant)\n"); + + /* Respect MIN_MUTE set by module parameters. */ + if (!(mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE)) + mixer->chip->quirk_flags |= QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL; + if (!(mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE)) + mixer->chip->quirk_flags |= QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL; + } else { + usb_audio_dbg(mixer->chip, "not applying MV-SILICON quirks on unknown variant"); + } +} + /* * Some Plantronics headsets have control names that don't meet ALSA naming * standards. This function fixes nonstandard source names. By the time @@ -4634,6 +4655,25 @@ triggered: usb_audio_dbg(chip, "something wrong in kctl name %s\n", id->name); } +static void snd_usb_mixer_fu_quirk_linear_scale(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + static const DECLARE_TLV_DB_LINEAR(scale, TLV_DB_GAIN_MUTE, 0); + + if (cval->min_mute) { + /* + * We are clearing SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + * resulting in min_mute being a no-op. + */ + usb_audio_warn(mixer->chip, "LINEAR_VOL overrides MIN_MUTE\n"); + } + + kctl->tlv.p = scale; + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; +} + void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, struct usb_mixer_elem_info *cval, int unitid, struct snd_kcontrol *kctl) @@ -4645,6 +4685,10 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, break; } + if (cval->control == UAC_FU_VOLUME && + !strncmp(mixer->chip->card->longname, "MV-SILICON", 10)) + snd_usb_mv_silicon_quirks(mixer, cval, kctl); + /* lowest playback value is muted on some devices */ if (mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE) if (strstr(kctl->id.name, "Playback")) { @@ -4660,6 +4704,21 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, "applying capture min mute quirk\n"); cval->min_mute = 1; } + + if (mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL) + if (cval->control == UAC_FU_VOLUME && strstr(kctl->id.name, "Playback")) { + usb_audio_info(mixer->chip, + "applying playback linear volume quirk\n"); + snd_usb_mixer_fu_quirk_linear_scale(mixer, cval, kctl); + } + + if (mixer->chip->quirk_flags & QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL) + if (cval->control == UAC_FU_VOLUME && strstr(kctl->id.name, "Capture")) { + usb_audio_info(mixer->chip, + "applying capture linear volume quirk\n"); + snd_usb_mixer_fu_quirk_linear_scale(mixer, cval, kctl); + } + /* ALSA-ify some Plantronics headset control names */ if (USB_ID_VENDOR(mixer->chip->usb_id) == 0x047f && (cval->control == UAC_FU_MUTE || cval->control == UAC_FU_VOLUME)) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 7eac7d1bce64..2e5a8d37ec57 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -362,6 +362,7 @@ static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, MIXER_LEVEL_0DB); break; + case USB_ID(0x194f, 0x0107): /* 1824 */ case USB_ID(0x194f, 0x010d): /* 1824c */ /* Set all output faders to unity gain */ a = SC1810C_SEL_OUTPUT; @@ -685,6 +686,7 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) return ret; break; + case USB_ID(0x194f, 0x0107): /* Presonus Studio 1824 */ case USB_ID(0x194f, 0x010d): /* Presonus Studio 1824c */ ret = snd_s1810c_switch_init(mixer, &snd_s1824c_mono_sw); if (ret < 0) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index fd1fb668929a..8eaa96222759 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2262,7 +2262,7 @@ static const struct scarlett2_device_entry scarlett2_devices[] = { { USB_ID(0x1235, 0x820c), &clarett_8pre_info, "Clarett+" }, /* End of list */ - { 0, NULL }, + { 0, NULL, NULL }, }; /* get the starting port index number for a given port type/direction */ diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c index 2ac813d57f4f..5f993b88448c 100644 --- a/sound/usb/qcom/qc_audio_offload.c +++ b/sound/usb/qcom/qc_audio_offload.c @@ -948,7 +948,7 @@ static int enable_audio_stream(struct snd_usb_substream *subs, _snd_pcm_hw_params_any(¶ms); m = hw_param_mask(¶ms, SNDRV_PCM_HW_PARAM_FORMAT); - snd_mask_leave(m, pcm_format); + snd_mask_leave(m, (__force unsigned int)pcm_format); i = hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_CHANNELS); snd_interval_setinteger(i); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index eafc0d73cca1..803e03d4d77b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2652,6 +2652,54 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * The AudioBox USB advertises S24_3LE as the only supported format + * for both playback and capture. It does not support S16_LE despite + * being a USB full-speed device. + */ + USB_DEVICE(0x194f, 0x0301), + QUIRK_DRIVER_INFO { + .vendor_name = "PreSonus", + .product_name = "AudioBox USB", + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { + QUIRK_DATA_AUDIOFORMAT(2) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + } + }, + { + QUIRK_DATA_AUDIOFORMAT(3) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + } + }, + QUIRK_COMPOSITE_END + } + } +}, #endif /* disabled */ { @@ -3900,5 +3948,70 @@ YAMAHA_DEVICE(0x7010, "UB99"), QUIRK_RME_DIGIFACE(0x3f8c), QUIRK_RME_DIGIFACE(0x3fa0), +#define QUIRK_AF16RIG(channel_count_, alt_setting_, \ + low_rate_, high_rate_, pack_size_, \ + clock_, interface_, endpoint_) \ + { \ + QUIRK_DATA_AUDIOFORMAT(interface_) { \ + .formats = SNDRV_PCM_FMTBIT_S32_LE, \ + .channels = channel_count_, \ + .fmt_type = UAC_FORMAT_TYPE_I_PCM, \ + .fmt_bits = 24, \ + .fmt_sz = 4, \ + .iface = interface_, \ + .altsetting = alt_setting_, \ + .altset_idx = alt_setting_, \ + .endpoint = endpoint_, \ + .ep_attr = USB_ENDPOINT_XFER_ISOC | \ + USB_ENDPOINT_SYNC_ASYNC, \ + .datainterval = 1, \ + .protocol = UAC_VERSION_2, \ + .maxpacksize = pack_size_, \ + .rates = SNDRV_PCM_RATE_##low_rate_ | \ + SNDRV_PCM_RATE_##high_rate_, \ + .rate_min = low_rate_, \ + .rate_max = high_rate_, \ + .nr_rates = 2, \ + .rate_table = (unsigned int[]) { \ + low_rate_, high_rate_ }, \ + .clock = clock_, \ + } \ + } + +#define QUIRK_AF16RIG_CLOCK(clock) \ + QUIRK_AF16RIG(34, 1, 44100, 48000, 0x3b8, clock, 1, 0x01), \ + QUIRK_AF16RIG(34, 1, 44100, 48000, 0x3b8, clock, 2, 0x81), \ + QUIRK_AF16RIG(18, 2, 88200, 96000, 0x3a8, clock, 1, 0x01), \ + QUIRK_AF16RIG(18, 2, 88200, 96000, 0x3a8, clock, 2, 0x81), \ + QUIRK_AF16RIG(10, 3, 176400, 192000, 0x3e8, clock, 1, 0x01), \ + QUIRK_AF16RIG(10, 3, 176400, 192000, 0x3e8, clock, 2, 0x81) + +/* Arturia AudioFuse 16Rig Audio */ +/* AF16Rig MIDI has USB PID 0xaf21 and appears to work OK without quirks */ +{ + USB_DEVICE(0x1c75, 0xaf20), + QUIRK_DRIVER_INFO { + .vendor_name = "Arturia", + .product_name = "AF16Rig", + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, + QUIRK_AF16RIG_CLOCK(41), /* Internal clock */ +#if 0 +/* These are disabled because I don't have the required hardware to test + * them. I suspect that the ADAT clock might not follow 176400 or 192000 + * because the AF16Rig won't accept ADAT audio data at those rates. + */ + QUIRK_AF16RIG_CLOCK(43), /* ADAT clock */ + QUIRK_AF16RIG_CLOCK(44), /* BNC word clock */ +#endif + { QUIRK_DATA_IGNORE(3) }, /* Firmware update */ + QUIRK_COMPOSITE_END + } + } +}, + +#undef QUIRK_AF16RIG_CLOCK +#undef QUIRK_AF16RIG + #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4cfa24c06fcd..519d9d1a2a41 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2,8 +2,11 @@ /* */ +#include <linux/cleanup.h> +#include <linux/err.h> #include <linux/init.h> #include <linux/slab.h> +#include <linux/string.h> #include <linux/usb.h> #include <linux/usb/audio.h> #include <linux/usb/midi.h> @@ -2135,16 +2138,69 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* * driver behavior quirk flags */ +struct usb_string_match { + const char *manufacturer; + const char *product; +}; + struct usb_audio_quirk_flags_table { u32 id; u32 flags; + const struct usb_string_match *usb_string_match; }; #define DEVICE_FLG(vid, pid, _flags) \ { .id = USB_ID(vid, pid), .flags = (_flags) } #define VENDOR_FLG(vid, _flags) DEVICE_FLG(vid, 0, _flags) +/* + * Use as a last resort if using DEVICE_FLG() is prone to VID/PID conflicts. + * + * Usage: + * // match vid, pid, "manufacturer", and "product" + * DEVICE_STRING_FLG(vid, pid, "manufacturer", "product", flags) + * + * // match vid, pid, "manufacturer", and any product string + * DEVICE_STRING_FLG(vid, pid, "manufacturer", NULL, flags) + * + * // match vid, pid, "manufacturer", and device must have no product string + * DEVICE_STRING_FLG(vid, pid, "manufacturer", "", flags) + * + * // match vid, pid, any manufacturer string, and "product" + * DEVICE_STRING_FLG(vid, pid, NULL, "product", flags) + * + * // match vid, pid, no manufacturer string, and "product" + * DEVICE_STRING_FLG(vid, pid, "", "product", flags) + * + * // match vid, pid, no manufacturer string, and no product string + * DEVICE_STRING_FLG(vid, pid, "", "", flags) + */ +#define DEVICE_STRING_FLG(vid, pid, _manufacturer, _product, _flags) \ +{ \ + .id = USB_ID(vid, pid), \ + .usb_string_match = &(const struct usb_string_match) { \ + .manufacturer = _manufacturer, \ + .product = _product, \ + }, \ + .flags = (_flags), \ +} + +/* + * Use as a last resort if using VENDOR_FLG() is prone to VID conflicts. + * + * Usage: + * // match vid, and "manufacturer" + * VENDOR_STRING_FLG(vid, "manufacturer", flags) + * + * // match vid, and device must have no manufacturer string + * VENDOR_STRING_FLG(vid, "", flags) + */ +#define VENDOR_STRING_FLG(vid, _manufacturer, _flags) \ + DEVICE_STRING_FLG(vid, 0, _manufacturer, NULL, _flags) + static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { + /* Device and string descriptor matches */ + /* Device matches */ DEVICE_FLG(0x001f, 0x0b21, /* AB13X USB Audio */ QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), @@ -2281,6 +2337,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x0d8c, 0x0014, /* C-Media */ QUIRK_FLAG_CTL_MSG_DELAY_1M | QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), + DEVICE_FLG(0x0e0b, 0xfa01, /* Feaulle Rainbow */ + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ @@ -2291,8 +2349,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x1101, 0x0003, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), - DEVICE_FLG(0x12d1, 0x3a07, /* Huawei Technologies Co., Ltd. */ - QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE | QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE), + DEVICE_FLG(0x12d1, 0x3a07, /* HUAWEI USB-C HEADSET */ + QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE | QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE | + QUIRK_FLAG_FORCE_IFACE_RESET | QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_MIC_RES_16), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ @@ -2421,6 +2480,13 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x84ef, 0x0082, /* Hotone Audio Pulze Mini */ + QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL | QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL), + + /* Vendor and string descriptor matches */ + VENDOR_STRING_FLG(0x1235, /* Conflict with Focusrite Novation */ + "MV-SILICON", + 0), /* Stop matching */ /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ @@ -2522,6 +2588,8 @@ static const char *const snd_usb_audio_quirk_flag_names[] = { QUIRK_STRING_ENTRY(MIXER_PLAYBACK_MIN_MUTE), QUIRK_STRING_ENTRY(MIXER_CAPTURE_MIN_MUTE), QUIRK_STRING_ENTRY(SKIP_IFACE_SETUP), + QUIRK_STRING_ENTRY(MIXER_PLAYBACK_LINEAR_VOL), + QUIRK_STRING_ENTRY(MIXER_CAPTURE_LINEAR_VOL), NULL }; @@ -2578,6 +2646,16 @@ void snd_usb_init_quirk_flags_table(struct snd_usb_audio *chip) if (chip->usb_id == p->id || (!USB_ID_PRODUCT(p->id) && USB_ID_VENDOR(chip->usb_id) == USB_ID_VENDOR(p->id))) { + /* Handle DEVICE_STRING_FLG/VENDOR_STRING_FLG. */ + if (p->usb_string_match && p->usb_string_match->manufacturer && + strcmp(p->usb_string_match->manufacturer, + chip->dev->manufacturer ? chip->dev->manufacturer : "")) + continue; + if (p->usb_string_match && p->usb_string_match->product && + strcmp(p->usb_string_match->product, + chip->dev->product ? chip->dev->product : "")) + continue; + snd_usb_apply_flag_dbg("builtin table", chip, p->flags); chip->quirk_flags |= p->flags; return; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d38c39e28f38..2532bf97e05e 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -366,6 +366,8 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor /* * TODO: this conversion is not complete, update it * after adding UAC3 values to asound.h + * NOTE: not all UAC3 channel relationship have a + * direct ALSA chmap equivalent. */ switch (is->bChRelationship) { case UAC3_CH_MONO: @@ -390,6 +392,12 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor case UAC3_CH_FRONT_RIGHT_OF_CENTER: map = SNDRV_CHMAP_FRC; break; + case UAC3_CH_FRONT_WIDE_LEFT: + map = SNDRV_CHMAP_FLW; + break; + case UAC3_CH_FRONT_WIDE_RIGHT: + map = SNDRV_CHMAP_FRW; + break; case UAC3_CH_SIDE_LEFT: map = SNDRV_CHMAP_SL; break; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 085530cf62d9..58fd07f8c3c9 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -228,6 +228,14 @@ extern bool snd_usb_skip_validation; * Skip the probe-time interface setup (usb_set_interface, * init_pitch, init_sample_rate); redundant with * snd_usb_endpoint_prepare() at stream-open time + * QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL + * Set linear volume mapping for devices where the playback volume control + * value is mapped to voltage (instead of dB) level linearly. In short: + * x(raw) = (raw - raw_min) / (raw_max - raw_min); V(x) = k * x; + * dB(x) = 20 * log10(x). Overrides QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE + * QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL + * Similar to QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL, but for capture streams. + * Overrides QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE */ enum { @@ -258,6 +266,8 @@ enum { QUIRK_TYPE_MIXER_PLAYBACK_MIN_MUTE = 24, QUIRK_TYPE_MIXER_CAPTURE_MIN_MUTE = 25, QUIRK_TYPE_SKIP_IFACE_SETUP = 26, + QUIRK_TYPE_MIXER_PLAYBACK_LINEAR_VOL = 27, + QUIRK_TYPE_MIXER_CAPTURE_LINEAR_VOL = 28, /* Please also edit snd_usb_audio_quirk_flag_names */ }; @@ -290,5 +300,7 @@ enum { #define QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE QUIRK_FLAG(MIXER_PLAYBACK_MIN_MUTE) #define QUIRK_FLAG_MIXER_CAPTURE_MIN_MUTE QUIRK_FLAG(MIXER_CAPTURE_MIN_MUTE) #define QUIRK_FLAG_SKIP_IFACE_SETUP QUIRK_FLAG(SKIP_IFACE_SETUP) +#define QUIRK_FLAG_MIXER_PLAYBACK_LINEAR_VOL QUIRK_FLAG(MIXER_PLAYBACK_LINEAR_VOL) +#define QUIRK_FLAG_MIXER_CAPTURE_LINEAR_VOL QUIRK_FLAG(MIXER_CAPTURE_LINEAR_VOL) #endif /* __USBAUDIO_H */ diff --git a/sound/usb/usx2y/us144mkii.c b/sound/usb/usx2y/us144mkii.c index 0cf4fa74e210..94553b61013c 100644 --- a/sound/usb/usx2y/us144mkii.c +++ b/sound/usb/usx2y/us144mkii.c @@ -420,7 +420,11 @@ static int tascam_probe(struct usb_interface *intf, /* The device has two interfaces; we drive both from this driver. */ if (intf->cur_altsetting->desc.bInterfaceNumber == 1) { - tascam = usb_get_intfdata(usb_ifnum_to_if(dev, 0)); + struct usb_interface *intf_zero = usb_ifnum_to_if(dev, 0); + + if (!intf_zero) + return -ENODEV; + tascam = usb_get_intfdata(intf_zero); if (tascam) { usb_set_intfdata(intf, tascam); tascam->iface1 = intf; diff --git a/tools/testing/selftests/alsa/utimer-test.c b/tools/testing/selftests/alsa/utimer-test.c index d221972cd8fb..1a9ff010cb11 100644 --- a/tools/testing/selftests/alsa/utimer-test.c +++ b/tools/testing/selftests/alsa/utimer-test.c @@ -15,6 +15,7 @@ #include <stdlib.h> #include <pthread.h> #include <string.h> +#include <errno.h> #define FRAME_RATE 8000 #define PERIOD_SIZE 4410 @@ -52,7 +53,14 @@ FIXTURE_SETUP(timer_f) { timer_dev_fd = open("/dev/snd/timer", O_RDONLY); ASSERT_GE(timer_dev_fd, 0); - ASSERT_EQ(ioctl(timer_dev_fd, SNDRV_TIMER_IOCTL_CREATE, self->utimer_info), 0); + if (ioctl(timer_dev_fd, SNDRV_TIMER_IOCTL_CREATE, self->utimer_info) < 0) { + int err = errno; + + close(timer_dev_fd); + if (err == ENOTTY || err == ENXIO) + SKIP(return, "CONFIG_SND_UTIMER not enabled"); + ASSERT_EQ(err, 0); + } ASSERT_GE(self->utimer_info->fd, 0); close(timer_dev_fd); |