diff options
Diffstat (limited to 'sound')
43 files changed, 444 insertions, 209 deletions
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7b977b753a03..7985dd8198b6 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, vendor_id); ret = device_add(&codec->dev); - if (ret) - goto err_free_codec; + if (ret) { + put_device(&codec->dev); + return ret; + } return 0; -err_free_codec: - of_node_put(codec->dev.of_node); - put_device(&codec->dev); - kfree(codec); - ac97_ctrl->codecs[idx] = NULL; - - return ret; } unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99b882158705..41905afada63 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; @@ -705,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: return -EPERM; + default: + break; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { snd_compr_drain_notify(stream); @@ -795,9 +807,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); if (retval) { @@ -817,6 +837,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; + /* next track doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) + return -EPERM; + /* you can signal next track if this is intended to be a gapless stream * and current track metadata is set */ @@ -834,9 +858,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) static int snd_compr_partial_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + + /* partial drain doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) return -EPERM; + /* stream can be drained only when next track has been signalled */ if (stream->next_track == false) return -EPERM; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 34390be3fb0f..11e653c8aa0e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -77,7 +77,7 @@ void snd_pcm_group_init(struct snd_pcm_group *group) spin_lock_init(&group->lock); mutex_init(&group->mutex); INIT_LIST_HEAD(&group->substreams); - refcount_set(&group->refs, 0); + refcount_set(&group->refs, 1); } /* define group lock helpers */ @@ -1096,8 +1096,7 @@ static void snd_pcm_group_unref(struct snd_pcm_group *group, if (!group) return; - do_free = refcount_dec_and_test(&group->refs) && - list_empty(&group->substreams); + do_free = refcount_dec_and_test(&group->refs); snd_pcm_group_unlock(group, substream->pcm->nonatomic); if (do_free) kfree(group); @@ -1874,6 +1873,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (!to_check) break; /* all drained */ init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); if (runtime->no_period_wakeup) @@ -1886,7 +1886,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } tout = msecs_to_jiffies(tout * 1000); } - tout = schedule_timeout_interruptible(tout); + tout = schedule_timeout(tout); snd_pcm_stream_lock_irq(substream); group = snd_pcm_stream_group_ref(substream); @@ -2020,6 +2020,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) snd_pcm_group_lock_irq(target_group, nonatomic); snd_pcm_stream_lock(substream1); snd_pcm_group_assign(substream1, target_group); + refcount_inc(&target_group->refs); snd_pcm_stream_unlock(substream1); snd_pcm_group_unlock_irq(target_group, nonatomic); _end: @@ -2056,13 +2057,14 @@ static int snd_pcm_unlink(struct snd_pcm_substream *substream) snd_pcm_group_lock_irq(group, nonatomic); relink_to_local(substream); + refcount_dec(&group->refs); /* detach the last stream, too */ if (list_is_singular(&group->substreams)) { relink_to_local(list_first_entry(&group->substreams, struct snd_pcm_substream, link_list)); - do_free = !refcount_read(&group->refs); + do_free = refcount_dec_and_test(&group->refs); } snd_pcm_group_unlock_irq(group, nonatomic); diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7737b2670064..6d9592f0ae1d 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1835,8 +1835,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr->type == USER_CLIENT) { info->input_pool = cptr->data.user.fifo_pool_size; info->input_free = info->input_pool; - if (cptr->data.user.fifo) - info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool); + info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo); } else { info->input_pool = 0; info->input_free = 0; diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index ea69261f269a..eaaa8b5830bb 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -263,3 +263,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return 0; } + +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) +{ + unsigned long flags; + int cells; + + if (!f) + return 0; + + snd_use_lock_use(&f->use_lock); + spin_lock_irqsave(&f->lock, flags); + cells = snd_seq_unused_cells(f->pool); + spin_unlock_irqrestore(&f->lock, flags); + snd_use_lock_free(&f->use_lock); + return cells; +} diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h index edc68743943d..b56a7b897c9c 100644 --- a/sound/core/seq/seq_fifo.h +++ b/sound/core/seq/seq_fifo.h @@ -53,5 +53,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table /* resize pool in fifo */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize); +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f); #endif diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9ea39348cdf5..7c6d1c277d4d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(hw_params); mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, rate, channels); if (err >= 0) ++oxfw->substreams_count; diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index 0d35359d25cd..0ecafd0c6722 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit, packets_per_page = PAGE_SIZE / packet_size; if (WARN_ON(!packets_per_page)) { err = -EINVAL; - goto error; + goto err_packets; } pages = DIV_ROUND_UP(count, packets_per_page); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 1192c7561d62..3c2db3816029 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -136,10 +136,12 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!acomp) return -ENODEV; if (!acomp->ops) { - request_module("i915"); - /* 60s timeout */ - wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(60 * 1000)); + if (!IS_ENABLED(CONFIG_MODULES) || + !request_module("i915")) { + /* 60s timeout */ + wait_for_completion_timeout(&bind_complete, + msecs_to_jiffies(60 * 1000)); + } } if (!acomp->ops) { dev_info(bus->dev, "couldn't bind with audio component\n"); diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 92390d457567..18e6546b4467 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -824,6 +824,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; + if (++depth > 10) + break; if (fix->chained_before) apply_fixup(codec, fix->chain_id, action, depth + 1); @@ -863,8 +865,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } if (!fix->chained || fix->chained_before) break; - if (++depth > 10) - break; id = fix->chain_id; } } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 133200d31170..a2fb19129219 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2948,7 +2948,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume; + bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ee5504e2441f..97a43a28b9e4 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; @@ -615,6 +613,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); + /* by some reason, the playback stream stalls on PulseAudio with + * tsched=1 when a capture stream triggers. Until we figure out the + * real cause, disable tsched mode by telling the PCM info flag. + */ + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) + runtime->hw.info |= SNDRV_PCM_INFO_BATCH; + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 146a71e0d594..82e26442724b 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -31,7 +31,7 @@ /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -/* 17 unused */ +#define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 485edaba0037..10d502328b76 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6009,7 +6009,8 @@ int snd_hda_gen_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); - snd_hda_apply_verbs(codec); + if (!spec->skip_verbs) + snd_hda_apply_verbs(codec); init_multi_out(codec); init_extra_out(codec); @@ -6051,6 +6052,24 @@ void snd_hda_gen_free(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_gen_free); +/** + * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting + * @codec: the HDA codec + * + * This can be put as patch_ops reboot_notify function. + */ +void snd_hda_gen_reboot_notify(struct hda_codec *codec) +{ + /* Make the codec enter D3 to avoid spurious noises from the internal + * speaker during (and after) reboot + */ + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify); + #ifdef CONFIG_PM /** * snd_hda_gen_check_power_status - check the loopback power save state @@ -6078,6 +6097,7 @@ static const struct hda_codec_ops generic_patch_ops = { .init = snd_hda_gen_init, .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, + .reboot_notify = snd_hda_gen_reboot_notify, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, #endif @@ -6100,7 +6120,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); if (err < 0) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 35a670a71c42..fb9f1a90238b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -243,6 +243,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */ /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; @@ -332,6 +333,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, struct auto_pin_cfg *cfg); int snd_hda_gen_build_controls(struct hda_codec *codec); int snd_hda_gen_build_pcms(struct hda_codec *codec); +void snd_hda_gen_reboot_notify(struct hda_codec *codec); /* standard jack event callbacks */ void snd_hda_gen_hp_automute(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6963dd852b5b..2d0db3c9f335 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -65,6 +65,7 @@ enum { POS_FIX_VIACOMBO, POS_FIX_COMBO, POS_FIX_SKL, + POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -137,7 +138,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -317,11 +318,10 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) -#define AZX_DCAPS_INTEL_BROXTON \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) +#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -337,6 +337,11 @@ enum { #define AZX_DCAPS_PRESET_ATI_HDMI_NS \ (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) +/* quirks for AMD SB */ +#define AZX_DCAPS_PRESET_AMD_SB \ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ @@ -846,6 +851,49 @@ static unsigned int azx_via_get_position(struct azx *chip, return bound_pos + mod_dma_pos; } +#define AMD_FIFO_SIZE 32 + +/* get the current DMA position with FIFO size correction */ +static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int pos, delay; + + pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); + if (!runtime) + return pos; + + runtime->delay = AMD_FIFO_SIZE; + delay = frames_to_bytes(runtime, AMD_FIFO_SIZE); + if (azx_dev->insufficient) { + if (pos < delay) { + delay = pos; + runtime->delay = bytes_to_frames(runtime, pos); + } else { + azx_dev->insufficient = 0; + } + } + + /* correct the DMA position for capture stream */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (pos < delay) + pos += azx_dev->core.bufsize; + pos -= delay; + } + + return pos; +} + +static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + + /* just read back the calculated value in the above */ + return substream->runtime->delay; +} + static unsigned int azx_skl_get_dpib_pos(struct azx *chip, struct azx_dev *azx_dev) { @@ -1422,6 +1470,7 @@ static int check_position_fix(struct azx *chip, int fix) case POS_FIX_VIACOMBO: case POS_FIX_COMBO: case POS_FIX_SKL: + case POS_FIX_FIFO: return fix; } @@ -1438,6 +1487,10 @@ static int check_position_fix(struct azx *chip, int fix) dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n"); return POS_FIX_VIACOMBO; } + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) { + dev_dbg(chip->card->dev, "Using FIFO position fix\n"); + return POS_FIX_FIFO; + } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { dev_dbg(chip->card->dev, "Using LPIB position fix\n"); return POS_FIX_LPIB; @@ -1458,6 +1511,7 @@ static void assign_position_fix(struct azx *chip, int fix) [POS_FIX_VIACOMBO] = azx_via_get_position, [POS_FIX_COMBO] = azx_get_pos_lpib, [POS_FIX_SKL] = azx_get_pos_skl, + [POS_FIX_FIFO] = azx_get_pos_fifo, }; chip->get_position[0] = chip->get_position[1] = callbacks[fix]; @@ -1472,6 +1526,9 @@ static void assign_position_fix(struct azx *chip, int fix) azx_get_delay_from_lpib; } + if (fix == POS_FIX_FIFO) + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_fifo; } /* @@ -2421,6 +2478,12 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD, X370 & co */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, + /* AMD, X570 & co */ + { PCI_DEVICE(0x1022, 0x1487), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0d51823d7270..6d1fb7c11f17 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1175,6 +1175,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ), SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ), SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI), diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f8d0845ee1e..968d3caab6ac 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -163,23 +163,10 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - switch (codec->core.vendor_id) { - case 0x14f12008: /* CX8200 */ - case 0x14f150f2: /* CX20722 */ - case 0x14f150f4: /* CX20724 */ - break; - default: - return; - } - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); - - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); + snd_hda_gen_reboot_notify(codec); } static void cx_auto_free(struct hda_codec *codec) @@ -624,18 +611,20 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec, /* update LED status via GPIO */ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + bool led_on) { struct conexant_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; if (spec->mute_led_polarity) - enabled = !enabled; + led_on = !led_on; - if (enabled) - spec->gpio_led &= ~mask; - else + if (led_on) spec->gpio_led |= mask; + else + spec->gpio_led &= ~mask; + codec_dbg(codec, "mask:%d enabled:%d gpio_led:%d\n", + mask, led_on, spec->gpio_led); if (spec->gpio_led != oldval) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_led); @@ -646,8 +635,8 @@ static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; struct conexant_spec *spec = codec->spec; - - cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + /* muted -> LED on */ + cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -669,7 +658,6 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 }, {} }; - codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led); if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook; @@ -1083,6 +1071,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de224cbea7a0..c1ddfd2fac52 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -837,9 +837,11 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + spec->gen.skip_verbs = 1; /* applied in below */ snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); + snd_hda_apply_verbs(codec); /* apply verbs here after own init */ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); @@ -869,15 +871,6 @@ static void alc_reboot_notify(struct hda_codec *codec) alc_shutup(codec); } -/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */ -static void alc_d3_at_reboot(struct hda_codec *codec) -{ - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); -} - #define alc_free snd_hda_gen_free #ifdef CONFIG_PM @@ -5152,7 +5145,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ + spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); @@ -5806,6 +5799,7 @@ enum { ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, + ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -6846,6 +6840,16 @@ static const struct hda_fixup alc269_fixups[] = { { } } }, + [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x411111f0 }, /* disable confusing internal speaker */ + { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6987,6 +6991,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7003,6 +7009,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), @@ -7080,6 +7087,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -8954,6 +8962,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 9ca9214cb7fb..5f40517717c4 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -10,7 +10,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_MAX98357A select SND_SOC_ADAU7002 select REGULATOR - depends on SND_SOC_AMD_ACP && I2C + depends on SND_SOC_AMD_ACP && I2C && GPIOLIB help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 9a406144b18f..befc2a3a05b0 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -674,8 +674,13 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, dev->channels = channels; ret = regmap_write(dev->regmap, MCHP_I2SMCC_MRA, mra); - if (ret < 0) + if (ret < 0) { + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } return ret; + } return regmap_write(dev->regmap, MCHP_I2SMCC_MRB, mrb); } @@ -690,31 +695,37 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream, err = wait_event_interruptible_timeout(dev->wq_txrdy, dev->tx_rdy, msecs_to_jiffies(500)); + if (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Tx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels)); + dev->tx_rdy = 1; + } } else { err = wait_event_interruptible_timeout(dev->wq_rxrdy, dev->rx_rdy, msecs_to_jiffies(500)); - } - - if (err == 0) { - u32 idra; - - dev_warn_once(dev->dev, "Timeout waiting for %s\n", - is_playback ? "Tx ready" : "Rx ready"); - if (is_playback) - idra = MCHP_I2SMCC_INT_TXRDY_MASK(dev->channels); - else - idra = MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels); - regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, idra); + if (err == 0) { + dev_warn_once(dev->dev, + "Timeout waiting for Rx ready\n"); + regmap_write(dev->regmap, MCHP_I2SMCC_IDRA, + MCHP_I2SMCC_INT_RXRDY_MASK(dev->channels)); + dev->rx_rdy = 1; + } } if (!mchp_i2s_mcc_is_running(dev)) { regmap_write(dev->regmap, MCHP_I2SMCC_CR, MCHP_I2SMCC_CR_CKDIS); if (dev->gclk_running) { - clk_disable_unprepare(dev->gclk); + clk_disable(dev->gclk); dev->gclk_running = 0; } + if (dev->gclk_use) { + clk_unprepare(dev->gclk); + dev->gclk_use = 0; + } } return 0; @@ -813,6 +824,8 @@ static int mchp_i2s_mcc_dai_probe(struct snd_soc_dai *dai) init_waitqueue_head(&dev->wq_txrdy); init_waitqueue_head(&dev->wq_rxrdy); + dev->tx_rdy = 1; + dev->rx_rdy = 1; snd_soc_dai_init_dma_data(dai, &dev->playback, &dev->capture); diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 9150e7068467..36eef1fb3d18 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -53,7 +53,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -91,7 +94,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 4, 0, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..be1e276e3631 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1628,14 +1628,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream, static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int reg_val = 0, reg_bclk_inv = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: reg_val |= RT1011_I2S_TDM_MS_S; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1645,7 +1649,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_bclk_inv |= RT1011_TDM_INV_BCLK; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1661,7 +1665,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_val |= RT1011_I2S_TDM_DF_PCM_B; break; default: - return -EINVAL; + ret = -EINVAL; } switch (dai->id) { @@ -1676,9 +1680,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); - return -EINVAL; + ret = -EINVAL; } - return 0; + + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_set_component_sysclk(struct snd_soc_component *component, @@ -1797,8 +1803,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int val = 0, tdm_en = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) tdm_en = RT1011_TDM_I2S_DOCK_EN_1; @@ -1818,7 +1828,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 2: break; default: - return -EINVAL; + ret = -EINVAL; } switch (slot_width) { @@ -1837,7 +1847,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 16: break; default: - return -EINVAL; + ret = -EINVAL; } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1854,7 +1864,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); - return 0; + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_probe(struct snd_soc_component *component) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 1186a03a88d6..6068bb697e22 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -223,6 +223,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, if (ipc->ops.reply_msg_match != NULL) header = ipc->ops.reply_msg_match(header, &mask); + else + mask = (u64)-1; if (list_empty(&ipc->rx_list)) { dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index 212370bf704c..3466675f2678 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -188,7 +188,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, memset(d->fw_read_buff, 0, FW_REG_BUF); if (w0_stat_sz > 0) - __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); + __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); for (offset = 0; offset < FW_REG_SIZE; offset += 16) { ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index ab3d23c7bd65..19f328d71f24 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -136,7 +136,7 @@ int skl_nhlt_update_topology_bin(struct skl_dev *skl) struct hdac_bus *bus = skl_to_bus(skl); struct device *dev = bus->dev; - dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", + dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", nhlt->header.oem_id, nhlt->header.oem_table_id, nhlt->header.oem_revision); diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index d16563408465..10ea4fdbeb1e 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; int hd_audio = 0; - int hd_align = 1; + int hd_align = 0; /* set hd mode */ switch (substream->runtime->format) { @@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S24_LE: hd_audio = 1; - hd_align = 0; break; default: dev_err(afe->dev, "%s() error: unsupported format %d\n", diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..d93db2c2b527 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -306,6 +306,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strncpy(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index b8690715abb5..aa9a1fca46fa 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -80,12 +80,6 @@ struct soc_tplg { static int soc_tplg_process_headers(struct soc_tplg *tplg); static void soc_tplg_complete(struct soc_tplg *tplg); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); -struct snd_soc_dapm_widget * -snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget); static void soc_tplg_denum_remove_texts(struct soc_enum *se); static void soc_tplg_denum_remove_values(struct soc_enum *se); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index f04d9fb5130f..d89b5c928c4d 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -187,57 +187,9 @@ static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); u32 spcr; u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - if (spcr & mask) { - /* start off disabled */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - spcr & ~mask); - toggle_clock(dev, playback); - } - if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { - /* Start the sample generator */ - spcr |= DAVINCI_MCBSP_SPCR_GRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - } - - if (playback) { - /* Stop the DMA to avoid data loss */ - /* while the transmitter is out of reset to handle XSYNCERR */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA stop failed\n"); - } - - /* Enable the transmitter */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr |= DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - - /* wait for any unexpected frame sync error to occur */ - udelay(100); - - /* Disable the transmitter to clear any outstanding XSYNCERR */ - spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - spcr &= ~DAVINCI_MCBSP_SPCR_XRST; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); - toggle_clock(dev, playback); - - /* Restart the DMA */ - if (component->driver->ops->trigger) { - int ret = component->driver->ops->trigger(substream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - printk(KERN_DEBUG "Playback DMA start failed\n"); - } - } /* Enable transmitter or receiver */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -575,7 +527,41 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 spcr; + u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; + davinci_mcbsp_stop(dev, playback); + + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (spcr & mask) { + /* start off disabled */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + spcr & ~mask); + toggle_clock(dev, playback); + } + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { + /* Start the sample generator */ + spcr |= DAVINCI_MCBSP_SPCR_GRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } + + if (playback) { + /* Enable the transmitter */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); + } + return 0; } diff --git a/sound/sound_core.c b/sound/sound_core.c index b730d97c4de6..90d118cd9164 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -275,7 +275,8 @@ retry: goto retry; } spin_unlock(&sound_loader_lock); - return -EBUSY; + r = -EBUSY; + goto fail; } } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 71d5f540334a..4c12cc5b53fd 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -72,7 +72,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) struct usb_host_endpoint *ep; ep = usb_pipe_endpoint(dev, pipe); - if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) return -EINVAL; return 0; } diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 14fc1e1d5d13..c406497c5919 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -600,14 +600,13 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) ret = hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, hiface_pcm_out_urb_handler); if (ret < 0) - return ret; + goto error; } ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); if (ret < 0) { - kfree(rt); dev_err(&chip->dev->dev, "Cannot create pcm instance\n"); - return ret; + goto error; } pcm->private_data = rt; @@ -620,4 +619,10 @@ int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) chip->pcm = rt; return 0; + +error: + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + kfree(rt); + return ret; } diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 2c03e0f6bf72..f70211e6b174 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -550,6 +550,15 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->volume_monitor = 255; line6pcm->line6 = line6; + spin_lock_init(&line6pcm->out.lock); + spin_lock_init(&line6pcm->in.lock); + line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; + + line6->line6pcm = line6pcm; + + pcm->private_data = line6pcm; + pcm->private_free = line6_cleanup_pcm; + line6pcm->max_packet_size_in = usb_maxpacket(line6->usbdev, usb_rcvisocpipe(line6->usbdev, ep_read), 0); @@ -562,15 +571,6 @@ int line6_init_pcm(struct usb_line6 *line6, return -EINVAL; } - spin_lock_init(&line6pcm->out.lock); - spin_lock_init(&line6pcm->in.lock); - line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; - - line6->line6pcm = line6pcm; - - pcm->private_data = line6pcm; - pcm->private_free = line6_cleanup_pcm; - err = line6_create_audio_out_urbs(line6pcm); if (err < 0) return err; diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index f0662bd4e50f..27bf61c177c0 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -368,7 +368,7 @@ static const struct line6_properties podhd_properties_table[] = { .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_HWMON, - .altsetting = 1, + .altsetting = 0, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0d24c72c155f..ed158f04de80 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -244,5 +244,5 @@ static struct usb_driver variax_driver = { module_usb_driver(variax_driver); -MODULE_DESCRIPTION("Vairax Workbench USB driver"); +MODULE_DESCRIPTION("Variax Workbench USB driver"); MODULE_LICENSE("GPL"); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 7498b5191b68..eceab19766db 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -68,6 +68,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -738,12 +739,13 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc) { int mu_channels; - void *c; if (desc->bLength < sizeof(*desc)) return -EINVAL; if (!desc->bNrInPins) return -EINVAL; + if (desc->bLength < sizeof(*desc) + desc->bNrInPins) + return -EINVAL; switch (state->mixer->protocol) { case UAC_VERSION_1: @@ -759,13 +761,6 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, break; } - if (!mu_channels) - return 0; - - c = uac_mixer_unit_bmControls(desc, state->mixer->protocol); - if (c - (void *)desc + (mu_channels - 1) / 8 >= desc->bLength) - return 0; /* no bmControls -> skip */ - return mu_channels; } @@ -773,16 +768,25 @@ static int uac_mixer_unit_get_channels(struct mixer_build *state, * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, +static int __check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { int protocol = state->mixer->protocol; int err; void *p1; + unsigned char *hdr; memset(term, 0, sizeof(*term)); - while ((p1 = find_audio_control_unit(state, id)) != NULL) { - unsigned char *hdr = p1; + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + + hdr = p1; term->id = id; if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { @@ -800,7 +804,7 @@ static int check_input_term(struct mixer_build *state, int id, /* call recursively to verify that the * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); + err = __check_input_term(state, d->bCSourceID, term); if (err < 0) return err; @@ -834,7 +838,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ @@ -897,7 +901,7 @@ static int check_input_term(struct mixer_build *state, int id, /* call recursively to verify that the * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); + err = __check_input_term(state, d->bCSourceID, term); if (err < 0) return err; @@ -948,7 +952,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC3_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; term->type = UAC3_SELECTOR_UNIT << 16; /* virtual type */ @@ -964,7 +968,7 @@ static int check_input_term(struct mixer_build *state, int id, return -EINVAL; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; @@ -982,6 +986,15 @@ static int check_input_term(struct mixer_build *state, int id, return -ENODEV; } + +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + memset(term, 0, sizeof(*term)); + memset(state->termbitmap, 0, sizeof(state->termbitmap)); + return __check_input_term(state, id, term); +} + /* * Feature Unit */ @@ -1988,6 +2001,31 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * Mixer Unit */ +/* check whether the given in/out overflows bmMixerControls matrix */ +static bool mixer_bitmap_overflow(struct uac_mixer_unit_descriptor *desc, + int protocol, int num_ins, int num_outs) +{ + u8 *hdr = (u8 *)desc; + u8 *c = uac_mixer_unit_bmControls(desc, protocol); + size_t rest; /* remaining bytes after bmMixerControls */ + + switch (protocol) { + case UAC_VERSION_1: + default: + rest = 1; /* iMixer */ + break; + case UAC_VERSION_2: + rest = 2; /* bmControls + iMixer */ + break; + case UAC_VERSION_3: + rest = 6; /* bmControls + wMixerDescrStr */ + break; + } + + /* overflow? */ + return c + (num_ins * num_outs + 7) / 8 + rest > hdr + hdr[0]; +} + /* * build a mixer unit control * @@ -2116,6 +2154,9 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, if (err < 0) return err; num_ins += iterm.channels; + if (mixer_bitmap_overflow(desc, state->mixer->protocol, + num_ins, num_outs)) + break; for (; ich < num_ins; ich++) { int och, ich_has_controls = 0; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199fa157a411..27dcb3743690 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1155,17 +1155,17 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + int unitid = 12; /* SampleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); - if (cval) { + if (mixer->id_elems[unitid]) { + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); + break; } - break; } } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 75b96929f76c..e4bbf79de956 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -339,6 +339,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; ifnum = 1; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7ee9d17d0143..e852c7fd6109 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1043,6 +1043,7 @@ found_clock: pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { + kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); return NULL; |