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authorMark Brown <broonie@kernel.org>2020-08-25 11:00:43 +0100
committerMark Brown <broonie@kernel.org>2020-08-25 11:00:43 +0100
commit1959ba4e40ce40c380dbe868433f5c4b20bb1ba3 (patch)
tree0bd58aa5433f8d4c207c058c259ec18fcba3e0f2 /sound
parent0235bc04627d02a08f7ad9d226a8fe78e6c4a1c3 (diff)
parentd012a7190fc1fd72ed48911e77ca97ba4521bccd (diff)
Merge tag 'v5.9-rc2' into asoc-5.9
Linux 5.9-rc2
Diffstat (limited to 'sound')
-rw-r--r--sound/atmel/ac97c.c20
-rw-r--r--sound/core/compress_offload.c4
-rw-r--r--sound/core/control_compat.c2
-rw-r--r--sound/core/info.c4
-rw-r--r--sound/core/init.c3
-rw-r--r--sound/core/memalloc.c9
-rw-r--r--sound/core/oss/pcm_oss.c2
-rw-r--r--sound/core/oss/pcm_plugin.c2
-rw-r--r--sound/core/pcm_iec958.c2
-rw-r--r--sound/core/pcm_memory.c1
-rw-r--r--sound/core/pcm_native.c10
-rw-r--r--sound/core/seq/oss/seq_oss.c8
-rw-r--r--sound/core/seq/oss/seq_oss_timer.c2
-rw-r--r--sound/core/seq/seq_midi_emul.c2
-rw-r--r--sound/core/sgbuf.c3
-rw-r--r--sound/core/vmaster.c263
-rw-r--r--sound/drivers/opl3/opl3_midi.c4
-rw-r--r--sound/drivers/opl3/opl3_synth.c2
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c2
-rw-r--r--sound/drivers/vx/vx_core.c3
-rw-r--r--sound/firewire/cmp.c1
-rw-r--r--sound/firewire/motu/motu-protocol-v3.c16
-rw-r--r--sound/hda/hdac_bus.c12
-rw-r--r--sound/hda/hdac_controller.c11
-rw-r--r--sound/hda/hdac_stream.c7
-rw-r--r--sound/isa/cs423x/cs4236_lib.c2
-rw-r--r--sound/isa/es18xx.c4
-rw-r--r--sound/isa/galaxy/galaxy.c6
-rw-r--r--sound/isa/gus/gus_reset.c2
-rw-r--r--sound/isa/gus/gus_uart.c3
-rw-r--r--sound/isa/msnd/msnd_pinnacle_mixer.c4
-rw-r--r--sound/isa/opti9xx/miro.c10
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c12
-rw-r--r--sound/isa/sb/sb16_csp.c2
-rw-r--r--sound/isa/sb/sb8_main.c10
-rw-r--r--sound/isa/sscape.c6
-rw-r--r--sound/oss/dmasound/dmasound_atari.c2
-rw-r--r--sound/oss/dmasound/dmasound_core.c4
-rw-r--r--sound/pci/ac97/ac97_codec.c4
-rw-r--r--sound/pci/ac97/ac97_patch.c34
-rw-r--r--sound/pci/asihpi/asihpi.c12
-rw-r--r--sound/pci/asihpi/hpi_internal.h2
-rw-r--r--sound/pci/asihpi/hpicmn.c26
-rw-r--r--sound/pci/atiixp.c6
-rw-r--r--sound/pci/au88x0/au88x0_a3ddata.c8
-rw-r--r--sound/pci/au88x0/au88x0_core.c12
-rw-r--r--sound/pci/au88x0/au88x0_xtalk.c36
-rw-r--r--sound/pci/aw2/aw2-saa7146.c2
-rw-r--r--sound/pci/azt3328.c2
-rw-r--r--sound/pci/bt87x.c14
-rw-r--r--sound/pci/ca0106/ca0106_mixer.c18
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c2
-rw-r--r--sound/pci/cs46xx/dsp_spos_scb_lib.c2
-rw-r--r--sound/pci/ctxfi/ctatc.c6
-rw-r--r--sound/pci/ctxfi/cthardware.c2
-rw-r--r--sound/pci/ctxfi/cthw20k1.c2
-rw-r--r--sound/pci/ctxfi/cthw20k2.c2
-rw-r--r--sound/pci/ctxfi/ctimap.c2
-rw-r--r--sound/pci/ctxfi/ctmixer.c2
-rw-r--r--sound/pci/ctxfi/ctpcm.c2
-rw-r--r--sound/pci/echoaudio/echoaudio.c188
-rw-r--r--sound/pci/echoaudio/echoaudio.h16
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c4
-rw-r--r--sound/pci/echoaudio/mona_dsp.c5
-rw-r--r--sound/pci/emu10k1/emu10k1_main.c4
-rw-r--r--sound/pci/emu10k1/emu10k1_patch.c3
-rw-r--r--sound/pci/emu10k1/emupcm.c5
-rw-r--r--sound/pci/es1938.c3
-rw-r--r--sound/pci/es1968.c20
-rw-r--r--sound/pci/fm801.c27
-rw-r--r--sound/pci/hda/Kconfig24
-rw-r--r--sound/pci/hda/hda_auto_parser.c6
-rw-r--r--sound/pci/hda/hda_beep.c2
-rw-r--r--sound/pci/hda/hda_codec.c106
-rw-r--r--sound/pci/hda/hda_controller.c11
-rw-r--r--sound/pci/hda/hda_controller.h4
-rw-r--r--sound/pci/hda/hda_generic.c156
-rw-r--r--sound/pci/hda/hda_generic.h15
-rw-r--r--sound/pci/hda/hda_intel.c70
-rw-r--r--sound/pci/hda/hda_local.h10
-rw-r--r--sound/pci/hda/hda_tegra.c4
-rw-r--r--sound/pci/hda/patch_ca0132.c22
-rw-r--r--sound/pci/hda/patch_conexant.c49
-rw-r--r--sound/pci/hda/patch_hdmi.c129
-rw-r--r--sound/pci/hda/patch_realtek.c462
-rw-r--r--sound/pci/hda/patch_sigmatel.c26
-rw-r--r--sound/pci/hda/thinkpad_helper.c19
-rw-r--r--sound/pci/ice1712/delta.c2
-rw-r--r--sound/pci/ice1712/juli.c20
-rw-r--r--sound/pci/ice1712/prodigy192.c2
-rw-r--r--sound/pci/ice1712/quartet.c14
-rw-r--r--sound/pci/intel8x0.c14
-rw-r--r--sound/pci/korg1212/korg1212.c4
-rw-r--r--sound/pci/mixart/mixart.c2
-rw-r--r--sound/pci/mixart/mixart_core.c2
-rw-r--r--sound/pci/nm256/nm256.c14
-rw-r--r--sound/pci/oxygen/oxygen_pcm.c2
-rw-r--r--sound/pci/oxygen/xonar_dg.c2
-rw-r--r--sound/pci/oxygen/xonar_wm87x6.c6
-rw-r--r--sound/pci/rme9652/hdspm.c4
-rw-r--r--sound/pci/via82xx.c8
-rw-r--r--sound/pci/via82xx_modem.c2
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c2
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.c1
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c4
-rw-r--r--sound/ppc/awacs.c12
-rw-r--r--sound/soc/codecs/cros_ec_codec.c27
-rw-r--r--sound/soc/fsl/fsl_sai.c5
-rw-r--r--sound/soc/fsl/fsl_sai.h2
-rw-r--r--sound/soc/meson/axg-card.c18
-rw-r--r--sound/soc/meson/gx-card.c18
-rw-r--r--sound/soc/meson/meson-card-utils.c4
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dai.c16
-rw-r--r--sound/soc/soc-pcm.c42
-rw-r--r--sound/soc/sof/probe.h8
-rw-r--r--sound/sparc/dbri.c10
-rw-r--r--sound/usb/6fire/control.c2
-rw-r--r--sound/usb/caiaq/audio.c2
-rw-r--r--sound/usb/caiaq/device.c2
-rw-r--r--sound/usb/card.c2
-rw-r--r--sound/usb/card.h7
-rw-r--r--sound/usb/clock.c2
-rw-r--r--sound/usb/endpoint.c25
-rw-r--r--sound/usb/format.c6
-rw-r--r--sound/usb/line6/capture.c2
-rw-r--r--sound/usb/line6/driver.c5
-rw-r--r--sound/usb/line6/driver.h8
-rw-r--r--sound/usb/line6/playback.c2
-rw-r--r--sound/usb/line6/podhd.c125
-rw-r--r--sound/usb/midi.c19
-rw-r--r--sound/usb/mixer.c27
-rw-r--r--sound/usb/mixer.h9
-rw-r--r--sound/usb/mixer_maps.c12
-rw-r--r--sound/usb/mixer_quirks.c4
-rw-r--r--sound/usb/mixer_s1810c.c6
-rw-r--r--sound/usb/mixer_scarlett_gen2.c4
-rw-r--r--sound/usb/mixer_us16x08.c2
-rw-r--r--sound/usb/pcm.c16
-rw-r--r--sound/usb/quirks-table.h171
-rw-r--r--sound/usb/quirks.c27
-rw-r--r--sound/usb/stream.c4
-rw-r--r--sound/xen/xen_snd_front.c6
-rw-r--r--sound/xen/xen_snd_front_evtchnl.c4
144 files changed, 1837 insertions, 986 deletions
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index a1dce9725b98..1006458f7f85 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -219,7 +219,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
switch (runtime->format) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
- case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+ case SNDRV_PCM_FORMAT_S16_BE:
word &= ~(AC97C_CMR_CEM_LITTLE);
break;
default:
@@ -301,7 +301,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
switch (runtime->format) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
- case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+ case SNDRV_PCM_FORMAT_S16_BE:
word &= ~(AC97C_CMR_CEM_LITTLE);
break;
default:
@@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd)
camr = ac97c_readl(chip, CAMR);
switch (cmd) {
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
- case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
ptcr = ATMEL_PDC_TXTEN;
camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX;
break;
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
- case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
ptcr |= ATMEL_PDC_TXTDIS;
if (chip->opened <= 1)
@@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd)
ptcr = readl(chip->regs + ATMEL_PDC_PTSR);
switch (cmd) {
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
- case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
ptcr = ATMEL_PDC_RXTEN;
camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX;
break;
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
- case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
ptcr |= ATMEL_PDC_RXTDIS;
if (chip->opened <= 1)
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 509290f2efa8..0e53f6f31916 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -764,6 +764,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
if (!retval) {
+ /* clear flags and stop any drain wait */
+ stream->partial_drain = false;
+ stream->metadata_set = false;
snd_compr_drain_notify(stream);
stream->runtime->total_bytes_available = 0;
stream->runtime->total_bytes_transferred = 0;
@@ -921,6 +924,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream)
if (stream->next_track == false)
return -EPERM;
+ stream->partial_drain = true;
retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
if (retval) {
pr_debug("Partial drain returned failure\n");
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index d55be1db1a8a..02df1d7db9a1 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -223,7 +223,7 @@ static int copy_ctl_value_from_user(struct snd_card *card,
{
struct snd_ctl_elem_value32 __user *data32 = userdata;
int i, type, size;
- int uninitialized_var(count);
+ int count;
unsigned int indirect;
if (copy_from_user(&data->id, &data32->id, sizeof(data->id)))
diff --git a/sound/core/info.c b/sound/core/info.c
index 8c6bc5241df5..9fec3070f8ba 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
{
int c;
- if (snd_BUG_ON(!buffer || !buffer->buffer))
+ if (snd_BUG_ON(!buffer))
+ return 1;
+ if (!buffer->buffer)
return 1;
if (len <= 0 || buffer->stop || buffer->error)
return 1;
diff --git a/sound/core/init.c b/sound/core/init.c
index b02a99766351..0478847ba2b8 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -203,7 +203,10 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
mutex_unlock(&snd_card_mutex);
card->dev = parent;
card->number = idx;
+#ifdef MODULE
+ WARN_ON(!module);
card->module = module;
+#endif
INIT_LIST_HEAD(&card->devices);
init_rwsem(&card->controls_rwsem);
rwlock_init(&card->ctl_files_rwlock);
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index bea46ed157a6..ad74ea9cbff5 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -135,16 +135,17 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
dmab->dev.type = type;
dmab->dev.dev = device;
dmab->bytes = 0;
+ dmab->area = NULL;
+ dmab->addr = 0;
+ dmab->private_data = NULL;
switch (type) {
case SNDRV_DMA_TYPE_CONTINUOUS:
gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL);
dmab->area = alloc_pages_exact(size, gfp);
- dmab->addr = 0;
break;
case SNDRV_DMA_TYPE_VMALLOC:
gfp = snd_mem_get_gfp_flags(device, GFP_KERNEL | __GFP_HIGHMEM);
dmab->area = __vmalloc(size, gfp);
- dmab->addr = 0;
break;
#ifdef CONFIG_HAS_DMA
#ifdef CONFIG_GENERIC_ALLOCATOR
@@ -157,7 +158,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
*/
dmab->dev.type = SNDRV_DMA_TYPE_DEV;
#endif /* CONFIG_GENERIC_ALLOCATOR */
- /* fall through */
+ fallthrough;
case SNDRV_DMA_TYPE_DEV:
case SNDRV_DMA_TYPE_DEV_UC:
snd_malloc_dev_pages(dmab, size);
@@ -171,8 +172,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
#endif
default:
pr_err("snd-malloc: invalid device type %d\n", type);
- dmab->area = NULL;
- dmab->addr = 0;
return -ENXIO;
}
if (! dmab->area)
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 68630244b00f..327ec42a36b0 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
if (substream)
break;
- /* Fall through */
+ fallthrough;
case VM_READ:
substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
break;
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 1545f8fdb4db..d5ca161d588c 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format,
if (snd_mask_test(format_mask, (__force int)format1))
return format1;
}
- /* fall through */
+ fallthrough;
default:
return (__force snd_pcm_format_t)-EINVAL;
}
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 073540f73b2f..f9a211cc1f2c 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -103,7 +103,7 @@ EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
/**
* snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
- * @hw_params: the hw_params instance for extracting rate and sample format
+ * @params: the hw_params instance for extracting rate and sample format
* @cs: channel status buffer, at least four bytes
* @len: length of channel status buffer
*
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 860935e3aea4..1bf6a3d9e0c2 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -39,6 +39,7 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev,
if (max_alloc_per_card &&
card->total_pcm_alloc_bytes + size > max_alloc_per_card)
return -ENOMEM;
+
err = snd_dma_alloc_pages(type, dev, size, dmab);
if (!err) {
mutex_lock(&card->memory_mutex);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 9630d2523948..9e0b2d73faf6 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream,
switch (substream->runtime->status->state) {
case SNDRV_PCM_STATE_PAUSED:
snd_pcm_pause(substream, false);
- /* fallthru */
+ fallthrough;
case SNDRV_PCM_STATE_SUSPENDED:
snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
break;
@@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream)
case SNDRV_PCM_STATE_DRAINING:
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
return -EBADFD;
- /* Fall through */
+ fallthrough;
case SNDRV_PCM_STATE_RUNNING:
return snd_pcm_update_hw_ptr(substream);
case SNDRV_PCM_STATE_PREPARED:
@@ -3713,7 +3713,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
area->vm_end - area->vm_start, area->vm_page_prot);
}
#endif /* CONFIG_GENERIC_ALLOCATOR */
-#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */
if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page &&
(substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV ||
substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_UC))
@@ -3722,7 +3721,6 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
substream->runtime->dma_area,
substream->runtime->dma_addr,
substream->runtime->dma_bytes);
-#endif /* CONFIG_X86 */
/* mmap with fault handler */
area->vm_ops = &snd_pcm_vm_ops_data_fault;
return 0;
@@ -3816,7 +3814,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD:
if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
return -ENXIO;
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW:
if (!pcm_status_mmap_allowed(pcm_file))
return -ENXIO;
@@ -3824,7 +3822,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area)
case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD:
if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT))
return -ENXIO;
- /* fallthrough */
+ fallthrough;
case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW:
if (!pcm_control_mmap_allowed(pcm_file))
return -ENXIO;
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 17f913657304..c8b9c0b315d8 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -168,10 +168,16 @@ static long
odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
{
struct seq_oss_devinfo *dp;
+ long rc;
+
dp = file->private_data;
if (snd_BUG_ON(!dp))
return -ENXIO;
- return snd_seq_oss_ioctl(dp, cmd, arg);
+
+ mutex_lock(&register_mutex);
+ rc = snd_seq_oss_ioctl(dp, cmd, arg);
+ mutex_unlock(&register_mutex);
+ return rc;
}
#ifdef CONFIG_COMPAT
diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c
index a35d429e4c27..f9f57232a83f 100644
--- a/sound/core/seq/oss/seq_oss_timer.c
+++ b/sound/core/seq/oss/seq_oss_timer.c
@@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev)
case TMR_WAIT_REL:
parm += rec->cur_tick;
rec->realtime = 0;
- /* fall through */
+ fallthrough;
case TMR_WAIT_ABS:
if (parm == 0) {
rec->realtime = 1;
diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c
index 198f285594e3..81d2ef5e5811 100644
--- a/sound/core/seq/seq_midi_emul.c
+++ b/sound/core/seq/seq_midi_emul.c
@@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv,
break;
case MIDI_CTL_MSB_DATA_ENTRY:
chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0;
- /* fall through */
+ fallthrough;
case MIDI_CTL_LSB_DATA_ENTRY:
if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED)
rpn(ops, drv, chan, chset);
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index c42217e2dd19..29ddb76187e5 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -142,6 +142,9 @@ unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
struct snd_sg_buf *sg = dmab->private_data;
unsigned int start, end, pg;
+ if (!sg)
+ return size;
+
start = ofs >> PAGE_SHIFT;
end = (ofs + size - 1) >> PAGE_SHIFT;
/* check page continuity */
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index ab54d79654c9..ab36f9898711 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
- * Virtual master and slave controls
+ * Virtual master and follower controls
*
* Copyright (c) 2008 by Takashi Iwai <tiwai@suse.de>
*/
@@ -21,15 +21,15 @@ struct link_ctl_info {
};
/*
- * link master - this contains a list of slave controls that are
+ * link master - this contains a list of follower controls that are
* identical types, i.e. info returns the same value type and value
* ranges, but may have different number of counts.
*
* The master control is so far only mono volume/switch for simplicity.
- * The same value will be applied to all slaves.
+ * The same value will be applied to all followers.
*/
struct link_master {
- struct list_head slaves;
+ struct list_head followers;
struct link_ctl_info info;
int val; /* the master value */
unsigned int tlv[4];
@@ -38,23 +38,23 @@ struct link_master {
};
/*
- * link slave - this contains a slave control element
+ * link follower - this contains a follower control element
*
- * It fakes the control callbacsk with additional attenuation by the
- * master control. A slave may have either one or two channels.
+ * It fakes the control callbacks with additional attenuation by the
+ * master control. A follower may have either one or two channels.
*/
-struct link_slave {
+struct link_follower {
struct list_head list;
struct link_master *master;
struct link_ctl_info info;
int vals[2]; /* current values */
unsigned int flags;
struct snd_kcontrol *kctl; /* original kcontrol pointer */
- struct snd_kcontrol slave; /* the copy of original control entry */
+ struct snd_kcontrol follower; /* the copy of original control entry */
};
-static int slave_update(struct link_slave *slave)
+static int follower_update(struct link_follower *follower)
{
struct snd_ctl_elem_value *uctl;
int err, ch;
@@ -62,68 +62,68 @@ static int slave_update(struct link_slave *slave)
uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
if (!uctl)
return -ENOMEM;
- uctl->id = slave->slave.id;
- err = slave->slave.get(&slave->slave, uctl);
+ uctl->id = follower->follower.id;
+ err = follower->follower.get(&follower->follower, uctl);
if (err < 0)
goto error;
- for (ch = 0; ch < slave->info.count; ch++)
- slave->vals[ch] = uctl->value.integer.value[ch];
+ for (ch = 0; ch < follower->info.count; ch++)
+ follower->vals[ch] = uctl->value.integer.value[ch];
error:
kfree(uctl);
return err < 0 ? err : 0;
}
-/* get the slave ctl info and save the initial values */
-static int slave_init(struct link_slave *slave)
+/* get the follower ctl info and save the initial values */
+static int follower_init(struct link_follower *follower)
{
struct snd_ctl_elem_info *uinfo;
int err;
- if (slave->info.count) {
+ if (follower->info.count) {
/* already initialized */
- if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE)
- return slave_update(slave);
+ if (follower->flags & SND_CTL_FOLLOWER_NEED_UPDATE)
+ return follower_update(follower);
return 0;
}
uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
if (!uinfo)
return -ENOMEM;
- uinfo->id = slave->slave.id;
- err = slave->slave.info(&slave->slave, uinfo);
+ uinfo->id = follower->follower.id;
+ err = follower->follower.info(&follower->follower, uinfo);
if (err < 0) {
kfree(uinfo);
return err;
}
- slave->info.type = uinfo->type;
- slave->info.count = uinfo->count;
- if (slave->info.count > 2 ||
- (slave->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
- slave->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
- pr_err("ALSA: vmaster: invalid slave element\n");
+ follower->info.type = uinfo->type;
+ follower->info.count = uinfo->count;
+ if (follower->info.count > 2 ||
+ (follower->info.type != SNDRV_CTL_ELEM_TYPE_INTEGER &&
+ follower->info.type != SNDRV_CTL_ELEM_TYPE_BOOLEAN)) {
+ pr_err("ALSA: vmaster: invalid follower element\n");
kfree(uinfo);
return -EINVAL;
}
- slave->info.min_val = uinfo->value.integer.min;
- slave->info.max_val = uinfo->value.integer.max;
+ follower->info.min_val = uinfo->value.integer.min;
+ follower->info.max_val = uinfo->value.integer.max;
kfree(uinfo);
- return slave_update(slave);
+ return follower_update(follower);
}
/* initialize master volume */
static int master_init(struct link_master *master)
{
- struct link_slave *slave;
+ struct link_follower *follower;
if (master->info.count)
return 0; /* already initialized */
- list_for_each_entry(slave, &master->slaves, list) {
- int err = slave_init(slave);
+ list_for_each_entry(follower, &master->followers, list) {
+ int err = follower_init(follower);
if (err < 0)
return err;
- master->info = slave->info;
+ master->info = follower->info;
master->info.count = 1; /* always mono */
/* set full volume as default (= no attenuation) */
master->val = master->info.max_val;
@@ -134,113 +134,113 @@ static int master_init(struct link_master *master)
return -ENOENT;
}
-static int slave_get_val(struct link_slave *slave,
- struct snd_ctl_elem_value *ucontrol)
+static int follower_get_val(struct link_follower *follower,
+ struct snd_ctl_elem_value *ucontrol)
{
int err, ch;
- err = slave_init(slave);
+ err = follower_init(follower);
if (err < 0)
return err;
- for (ch = 0; ch < slave->info.count; ch++)
- ucontrol->value.integer.value[ch] = slave->vals[ch];
+ for (ch = 0; ch < follower->info.count; ch++)
+ ucontrol->value.integer.value[ch] = follower->vals[ch];
return 0;
}
-static int slave_put_val(struct link_slave *slave,
- struct snd_ctl_elem_value *ucontrol)
+static int follower_put_val(struct link_follower *follower,
+ struct snd_ctl_elem_value *ucontrol)
{
int err, ch, vol;
- err = master_init(slave->master);
+ err = master_init(follower->master);
if (err < 0)
return err;
- switch (slave->info.type) {
+ switch (follower->info.type) {
case SNDRV_CTL_ELEM_TYPE_BOOLEAN:
- for (ch = 0; ch < slave->info.count; ch++)
+ for (ch = 0; ch < follower->info.count; ch++)
ucontrol->value.integer.value[ch] &=
- !!slave->master->val;
+ !!follower->master->val;
break;
case SNDRV_CTL_ELEM_TYPE_INTEGER:
- for (ch = 0; ch < slave->info.count; ch++) {
+ for (ch = 0; ch < follower->info.count; ch++) {
/* max master volume is supposed to be 0 dB */
vol = ucontrol->value.integer.value[ch];
- vol += slave->master->val - slave->master->info.max_val;
- if (vol < slave->info.min_val)
- vol = slave->info.min_val;
- else if (vol > slave->info.max_val)
- vol = slave->info.max_val;
+ vol += follower->master->val - follower->master->info.max_val;
+ if (vol < follower->info.min_val)
+ vol = follower->info.min_val;
+ else if (vol > follower->info.max_val)
+ vol = follower->info.max_val;
ucontrol->value.integer.value[ch] = vol;
}
break;
}
- return slave->slave.put(&slave->slave, ucontrol);
+ return follower->follower.put(&follower->follower, ucontrol);
}
/*
- * ctl callbacks for slaves
+ * ctl callbacks for followers
*/
-static int slave_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
+static int follower_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- return slave->slave.info(&slave->slave, uinfo);
+ struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+ return follower->follower.info(&follower->follower, uinfo);
}
-static int slave_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int follower_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- return slave_get_val(slave, ucontrol);
+ struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+ return follower_get_val(follower, ucontrol);
}
-static int slave_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int follower_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ struct link_follower *follower = snd_kcontrol_chip(kcontrol);
int err, ch, changed = 0;
- err = slave_init(slave);
+ err = follower_init(follower);
if (err < 0)
return err;
- for (ch = 0; ch < slave->info.count; ch++) {
- if (slave->vals[ch] != ucontrol->value.integer.value[ch]) {
+ for (ch = 0; ch < follower->info.count; ch++) {
+ if (follower->vals[ch] != ucontrol->value.integer.value[ch]) {
changed = 1;
- slave->vals[ch] = ucontrol->value.integer.value[ch];
+ follower->vals[ch] = ucontrol->value.integer.value[ch];
}
}
if (!changed)
return 0;
- err = slave_put_val(slave, ucontrol);
+ err = follower_put_val(follower, ucontrol);
if (err < 0)
return err;
return 1;
}
-static int slave_tlv_cmd(struct snd_kcontrol *kcontrol,
- int op_flag, unsigned int size,
- unsigned int __user *tlv)
+static int follower_tlv_cmd(struct snd_kcontrol *kcontrol,
+ int op_flag, unsigned int size,
+ unsigned int __user *tlv)
{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
+ struct link_follower *follower = snd_kcontrol_chip(kcontrol);
/* FIXME: this assumes that the max volume is 0 dB */
- return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv);
+ return follower->follower.tlv.c(&follower->follower, op_flag, size, tlv);
}
-static void slave_free(struct snd_kcontrol *kcontrol)
+static void follower_free(struct snd_kcontrol *kcontrol)
{
- struct link_slave *slave = snd_kcontrol_chip(kcontrol);
- if (slave->slave.private_free)
- slave->slave.private_free(&slave->slave);
- if (slave->master)
- list_del(&slave->list);
- kfree(slave);
+ struct link_follower *follower = snd_kcontrol_chip(kcontrol);
+ if (follower->follower.private_free)
+ follower->follower.private_free(&follower->follower);
+ if (follower->master)
+ list_del(&follower->list);
+ kfree(follower);
}
/*
- * Add a slave control to the group with the given master control
+ * Add a follower control to the group with the given master control
*
- * All slaves must be the same type (returning the same information
+ * All followers must be the same type (returning the same information
* via info callback). The function doesn't check it, so it's your
* responsibility.
*
@@ -249,35 +249,36 @@ static void slave_free(struct snd_kcontrol *kcontrol)
* - logarithmic volume control (dB level), no linear volume
* - master can only attenuate the volume, no gain
*/
-int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
- unsigned int flags)
+int _snd_ctl_add_follower(struct snd_kcontrol *master,
+ struct snd_kcontrol *follower,
+ unsigned int flags)
{
struct link_master *master_link = snd_kcontrol_chip(master);
- struct link_slave *srec;
+ struct link_follower *srec;
- srec = kzalloc(struct_size(srec, slave.vd, slave->count),
+ srec = kzalloc(struct_size(srec, follower.vd, follower->count),
GFP_KERNEL);
if (!srec)
return -ENOMEM;
- srec->kctl = slave;
- srec->slave = *slave;
- memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
+ srec->kctl = follower;
+ srec->follower = *follower;
+ memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd));
srec->master = master_link;
srec->flags = flags;
/* override callbacks */
- slave->info = slave_info;
- slave->get = slave_get;
- slave->put = slave_put;
- if (slave->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
- slave->tlv.c = slave_tlv_cmd;
- slave->private_data = srec;
- slave->private_free = slave_free;
-
- list_add_tail(&srec->list, &master_link->slaves);
+ follower->info = follower_info;
+ follower->get = follower_get;
+ follower->put = follower_put;
+ if (follower->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)
+ follower->tlv.c = follower_tlv_cmd;
+ follower->private_data = srec;
+ follower->private_free = follower_free;
+
+ list_add_tail(&srec->list, &master_link->followers);
return 0;
}
-EXPORT_SYMBOL(_snd_ctl_add_slave);
+EXPORT_SYMBOL(_snd_ctl_add_follower);
/*
* ctl callbacks for master controls
@@ -309,20 +310,20 @@ static int master_get(struct snd_kcontrol *kcontrol,
return 0;
}
-static int sync_slaves(struct link_master *master, int old_val, int new_val)
+static int sync_followers(struct link_master *master, int old_val, int new_val)
{
- struct link_slave *slave;
+ struct link_follower *follower;
struct snd_ctl_elem_value *uval;
uval = kmalloc(sizeof(*uval), GFP_KERNEL);
if (!uval)
return -ENOMEM;
- list_for_each_entry(slave, &master->slaves, list) {
+ list_for_each_entry(follower, &master->followers, list) {
master->val = old_val;
- uval->id = slave->slave.id;
- slave_get_val(slave, uval);
+ uval->id = follower->follower.id;
+ follower_get_val(follower, uval);
master->val = new_val;
- slave_put_val(slave, uval);
+ follower_put_val(follower, uval);
}
kfree(uval);
return 0;
@@ -344,7 +345,7 @@ static int master_put(struct snd_kcontrol *kcontrol,
if (new_val == old_val)
return 0;
- err = sync_slaves(master, old_val, new_val);
+ err = sync_followers(master, old_val, new_val);
if (err < 0)
return err;
if (master->hook && !first_init)
@@ -355,17 +356,17 @@ static int master_put(struct snd_kcontrol *kcontrol,
static void master_free(struct snd_kcontrol *kcontrol)
{
struct link_master *master = snd_kcontrol_chip(kcontrol);
- struct link_slave *slave, *n;
+ struct link_follower *follower, *n;
- /* free all slave links and retore the original slave kctls */
- list_for_each_entry_safe(slave, n, &master->slaves, list) {
- struct snd_kcontrol *sctl = slave->kctl;
+ /* free all follower links and retore the original follower kctls */
+ list_for_each_entry_safe(follower, n, &master->followers, list) {
+ struct snd_kcontrol *sctl = follower->kctl;
struct list_head olist = sctl->list;
- memcpy(sctl, &slave->slave, sizeof(*sctl));
- memcpy(sctl->vd, slave->slave.vd,
+ memcpy(sctl, &follower->follower, sizeof(*sctl));
+ memcpy(sctl->vd, follower->follower.vd,
sctl->count * sizeof(*sctl->vd));
sctl->list = olist; /* keep the current linked-list */
- kfree(slave);
+ kfree(follower);
}
kfree(master);
}
@@ -378,8 +379,8 @@ static void master_free(struct snd_kcontrol *kcontrol)
*
* Creates a virtual master control with the given name string.
*
- * After creating a vmaster element, you can add the slave controls
- * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached().
+ * After creating a vmaster element, you can add the follower controls
+ * via snd_ctl_add_follower() or snd_ctl_add_follower_uncached().
*
* The optional argument @tlv can be used to specify the TLV information
* for dB scale of the master control. It should be a single element
@@ -403,7 +404,7 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
master = kzalloc(sizeof(*master), GFP_KERNEL);
if (!master)
return NULL;
- INIT_LIST_HEAD(&master->slaves);
+ INIT_LIST_HEAD(&master->followers);
kctl = snd_ctl_new1(&knew, master);
if (!kctl) {
@@ -455,11 +456,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
/**
- * snd_ctl_sync_vmaster - Sync the vmaster slaves and hook
+ * snd_ctl_sync_vmaster - Sync the vmaster followers and hook
* @kcontrol: vmaster kctl element
* @hook_only: sync only the hook
*
- * Forcibly call the put callback of each slave and call the hook function
+ * Forcibly call the put callback of each follower and call the hook function
* to synchronize with the current value of the given vmaster element.
* NOP when NULL is passed to @kcontrol.
*/
@@ -476,7 +477,7 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
if (err < 0)
return;
first_init = err;
- err = sync_slaves(master, master->val, master->val);
+ err = sync_followers(master, master->val, master->val);
if (err < 0)
return;
}
@@ -487,34 +488,34 @@ void snd_ctl_sync_vmaster(struct snd_kcontrol *kcontrol, bool hook_only)
EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster);
/**
- * snd_ctl_apply_vmaster_slaves - Apply function to each vmaster slave
+ * snd_ctl_apply_vmaster_followers - Apply function to each vmaster follower
* @kctl: vmaster kctl element
* @func: function to apply
* @arg: optional function argument
*
- * Apply the function @func to each slave kctl of the given vmaster kctl.
+ * Apply the function @func to each follower kctl of the given vmaster kctl.
* Returns 0 if successful, or a negative error code.
*/
-int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
- int (*func)(struct snd_kcontrol *vslave,
- struct snd_kcontrol *slave,
- void *arg),
- void *arg)
+int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
+ int (*func)(struct snd_kcontrol *vfollower,
+ struct snd_kcontrol *follower,
+ void *arg),
+ void *arg)
{
struct link_master *master;
- struct link_slave *slave;
+ struct link_follower *follower;
int err;
master = snd_kcontrol_chip(kctl);
err = master_init(master);
if (err < 0)
return err;
- list_for_each_entry(slave, &master->slaves, list) {
- err = func(slave->kctl, &slave->slave, arg);
+ list_for_each_entry(follower, &master->followers, list) {
+ err = func(follower->kctl, &follower->follower, arg);
if (err < 0)
return err;
}
return 0;
}
-EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_slaves);
+EXPORT_SYMBOL_GPL(snd_ctl_apply_vmaster_followers);
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 2f6e8023e05c..eb23c55323ae 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
instr_4op = 1;
break;
}
- /* fall through */
+ fallthrough;
default:
spin_unlock_irqrestore(&opl3->voice_lock, flags);
return;
@@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
switch (connection) {
case 0x03:
snd_opl3_calc_volume(&vol_op[2], vel, chan);
- /* fallthru */
+ fallthrough;
case 0x02:
snd_opl3_calc_volume(&vol_op[0], vel, chan);
break;
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index e69a4ef0d6bd..08c10ac9d6c8 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
{
struct snd_dm_fm_info info;
+ memset(&info, 0, sizeof(info));
+
info.fm_mode = opl3->fm_mode;
info.rhythm = opl3->rhythm;
if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 05244b11ed5e..4e79293d7f11 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -36,7 +36,7 @@ static void pcsp_call_pcm_elapsed(unsigned long priv)
}
}
-static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0);
+static DECLARE_TASKLET_OLD(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed);
/* write the port and returns the next expire time in ns;
* called at the trigger-start and in hrtimer callback
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index ffab0400d7fb..26d591fe6a6b 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -511,8 +511,9 @@ irqreturn_t snd_vx_threaded_irq_handler(int irq, void *dev)
/* The start on time code conditions are filled (ie the time code
* received by the board is equal to one of those given to it).
*/
- if (events & TIME_CODE_EVENT_PENDING)
+ if (events & TIME_CODE_EVENT_PENDING) {
; /* so far, nothing to do yet */
+ }
/* The frequency has changed on the board (UER mode). */
if (events & FREQUENCY_CHANGE_EVENT_PENDING)
diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c
index 14abbe7175b6..b596bec19774 100644
--- a/sound/firewire/cmp.c
+++ b/sound/firewire/cmp.c
@@ -293,7 +293,6 @@ static int pcr_set_check(struct cmp_connection *c, __be32 pcr)
/**
* cmp_connection_establish - establish a connection to the target
* @c: the connection manager
- * @max_payload_bytes: the amount of data (including CIP headers) per packet
*
* This function establishes a point-to-point connection from the local
* computer to the target by allocating isochronous resources (channel and
diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c
index 01a47ac7bb2d..4e6b0e449ee4 100644
--- a/sound/firewire/motu/motu-protocol-v3.c
+++ b/sound/firewire/motu/motu-protocol-v3.c
@@ -24,6 +24,9 @@
#define V3_NO_ADAT_OPT_OUT_IFACE_A 0x00040000
#define V3_NO_ADAT_OPT_OUT_IFACE_B 0x00400000
+#define V3_MSG_FLAG_CLK_CHANGED 0x00000002
+#define V3_CLK_WAIT_MSEC 4000
+
int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu,
unsigned int *rate)
{
@@ -79,9 +82,16 @@ int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu,
return err;
if (need_to_wait) {
- /* Cost expensive. */
- if (msleep_interruptible(4000) > 0)
- return -EINTR;
+ int result;
+
+ motu->msg = 0;
+ result = wait_event_interruptible_timeout(motu->hwdep_wait,
+ motu->msg & V3_MSG_FLAG_CLK_CHANGED,
+ msecs_to_jiffies(V3_CLK_WAIT_MSEC));
+ if (result < 0)
+ return result;
+ if (result == 0)
+ return -ETIMEDOUT;
}
return 0;
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 09ddab5f5cae..9766f6af8743 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev,
INIT_LIST_HEAD(&bus->hlink_list);
init_waitqueue_head(&bus->rirb_wq);
bus->irq = -1;
+
+ /*
+ * Default value of '8' is as per the HD audio specification (Rev 1.0a).
+ * Following relation is used to derive STRIPE control value.
+ * For sample rate <= 48K:
+ * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
+ * For sample rate > 48K:
+ * { ((num_channels * bits_per_sample * rate/48000) /
+ * number of SDOs) >= 8 }
+ */
+ bus->sdo_limit = 8;
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init);
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 011b17cc1efa..b98449fd92f3 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset)
bus->chip_init = true;
- /*
- * Default value of '8' is as per the HD audio specification (Rev 1.0a).
- * Following relation is used to derive STRIPE control value.
- * For sample rate <= 48K:
- * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
- * For sample rate > 48K:
- * { ((num_channels * bits_per_sample * rate/48000) /
- * number of SDOs) >= 8 }
- */
- bus->sdo_limit = 8;
-
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index a38a2af1654f..abe7a1b16fe1 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -150,9 +150,12 @@ void snd_hdac_stream_reset(struct hdac_stream *azx_dev)
{
unsigned char val;
int timeout;
+ int dma_run_state;
snd_hdac_stream_clear(azx_dev);
+ dma_run_state = snd_hdac_stream_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START;
+
snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET);
udelay(3);
timeout = 300;
@@ -162,6 +165,10 @@ void snd_hdac_stream_reset(struct hdac_stream *azx_dev)
if (val)
break;
} while (--timeout);
+
+ if (azx_dev->bus->dma_stop_delay && dma_run_state)
+ udelay(azx_dev->bus->dma_stop_delay);
+
val &= ~SD_CTL_STREAM_RESET;
snd_hdac_stream_writeb(azx_dev, SD_CTL, val);
udelay(3);
diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c
index 4a028f42bb74..52f05adb1870 100644
--- a/sound/isa/cs423x/cs4236_lib.c
+++ b/sound/isa/cs423x/cs4236_lib.c
@@ -39,7 +39,7 @@
* D7: consumer serial port enable (CS4237B,CS4238B)
* D6: channels status block reset (CS4237B,CS4238B)
* D5: user bit in sub-frame of digital audio data (CS4237B,CS4238B)
- * D4: validity bit bit in sub-frame of digital audio data (CS4237B,CS4238B)
+ * D4: validity bit in sub-frame of digital audio data (CS4237B,CS4238B)
*
* C5 lower channel status (digital serial data description) (CS4237B,CS4238B)
* D7-D6: first two bits of category code
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index d1135f6ae104..5f8d7e8a5477 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -955,7 +955,7 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
case 0x1887:
case 0x1888:
return snd_ctl_enum_info(uinfo, 1, 5, texts5Source);
- case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */
+ case 0x1869: /* DS somewhat contradictory for 1869: could be 5 or 8 */
case 0x1879:
return snd_ctl_enum_info(uinfo, 1, 8, texts8Source);
default:
@@ -998,7 +998,7 @@ static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
val = 3;
} else
retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00;
- /* fall through */
+ fallthrough;
/* 4 source chips */
case 0x1868:
case 0x1878:
diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c
index ce409e75ae51..65f9f46c9f58 100644
--- a/sound/isa/galaxy/galaxy.c
+++ b/sound/isa/galaxy/galaxy.c
@@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
break;
case 2:
irq[n] = 9;
- /* Fall through */
+ fallthrough;
case 9:
wss_config[n] |= WSS_CONFIG_IRQ_9;
break;
@@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n)
case 1:
if (dma1[n] == 0)
break;
- /* Fall through */
+ fallthrough;
default:
dev_err(dev, "invalid capture DMA %d\n", dma2[n]);
return 0;
@@ -322,7 +322,7 @@ mpu:
break;
case 2:
mpu_irq[n] = 9;
- /* Fall through */
+ fallthrough;
case 9:
config[n] |= GALAXY_CONFIG_MPUIRQ_2;
break;
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
index 07bfcda43827..9a1ab5872c4f 100644
--- a/sound/isa/gus/gus_reset.c
+++ b/sound/isa/gus/gus_reset.c
@@ -9,8 +9,6 @@
#include <sound/core.h>
#include <sound/gus.h>
-extern void snd_gf1_timers_init(struct snd_gus_card * gus);
-extern void snd_gf1_timers_done(struct snd_gus_card * gus);
extern int snd_gf1_synth_init(struct snd_gus_card * gus);
extern void snd_gf1_synth_done(struct snd_gus_card * gus);
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
index 7586619770b3..4fb4ed79e262 100644
--- a/sound/isa/gus/gus_uart.c
+++ b/sound/isa/gus/gus_uart.c
@@ -13,7 +13,8 @@
static void snd_gf1_interrupt_midi_in(struct snd_gus_card * gus)
{
int count;
- unsigned char stat, data, byte;
+ unsigned char stat, byte;
+ __always_unused unsigned char data;
unsigned long flags;
count = 10;
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 02c566fca9e5..63633bd41e5b 100644
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -219,11 +219,9 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right)
case MSND_MIXER_VOLUME: /* master volume */
writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft);
writew(wRight, dev->SMA + SMA_wCurrMastVolRight);
- /* fall through */
-
+ fallthrough;
case MSND_MIXER_AUX: /* aux pot control */
/* scaled by master volume */
- /* fall through */
/* digital controls */
case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index b039429e6871..44ed1b65f6ce 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci)
switch (timeout-ACI_MINTIME) {
case 0 ... 9:
out /= 10;
- /* fall through */
+ fallthrough;
case 10 ... 19:
out /= 10;
- /* fall through */
+ fallthrough;
case 20 ... 30:
out /= 10;
- /* fall through */
+ fallthrough;
default:
set_current_state(TASK_UNINTERRUPTIBLE);
schedule_timeout(out);
@@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip,
retval = inb(chip->mc_base + 9);
break;
}
- /* fall through */
+ fallthrough;
case OPTi9XX_HW_82C929:
retval = inb(chip->mc_base + reg);
@@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg,
outb(value, chip->mc_base + 9);
break;
}
- /* fall through */
+ fallthrough;
case OPTi9XX_HW_82C929:
outb(value, chip->mc_base + reg);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 0e6d20e49158..881d3b5711d2 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip,
retval = inb(chip->mc_base + 9);
break;
}
- /* Fall through */
+ fallthrough;
case OPTi9XX_HW_82C928:
case OPTi9XX_HW_82C929:
@@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
outb(value, chip->mc_base + 9);
break;
}
- /* Fall through */
+ fallthrough;
case OPTi9XX_HW_82C928:
case OPTi9XX_HW_82C929:
@@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc);
/* enable wave audio */
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02);
- /* Fall through */
+ fallthrough;
case OPTi9XX_HW_82C925:
/* enable WSS mode */
@@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
case OPTi9XX_HW_82C931:
/* disable 3D sound (set GPIO1 as output, low) */
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
- /* fall through */
+ fallthrough;
+
case OPTi9XX_HW_82C933:
/*
* The BTC 1817DW has QS1000 wavetable which is connected
@@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip,
* or digital input signal.
*/
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01);
- /* fall through */
+ fallthrough;
+
case OPTi9XX_HW_82C930:
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03);
snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff);
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 4ad0ff0c4508..270af863e198 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -102,7 +102,7 @@ static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buff
int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep)
{
struct snd_sb_csp *p;
- int uninitialized_var(version);
+ int version;
int err;
struct snd_hwdep *hw;
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index e33dfe165276..86d0d2fdf48a 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream)
chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
break;
}
- /* fall through */
+ fallthrough;
case SB_HW_201:
if (rate > 23000) {
chip->playback_format = SB_DSP_HI_OUTPUT_AUTO;
break;
}
- /* fall through */
+ fallthrough;
case SB_HW_20:
chip->playback_format = SB_DSP_LO_OUTPUT_AUTO;
break;
@@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
chip->capture_format = SB_DSP_HI_INPUT_AUTO;
break;
}
- /* fall through */
+ fallthrough;
case SB_HW_20:
chip->capture_format = SB_DSP_LO_INPUT_AUTO;
break;
@@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
case SB_MODE_PLAYBACK_16: /* ok.. playback is active */
if (chip->hardware != SB_HW_JAZZ16)
break;
- /* fall through */
+ fallthrough;
case SB_MODE_PLAYBACK_8:
substream = chip->playback_substream;
if (chip->playback_format == SB_DSP_OUTPUT)
@@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip)
case SB_MODE_CAPTURE_16:
if (chip->hardware != SB_HW_JAZZ16)
break;
- /* fall through */
+ fallthrough;
case SB_MODE_CAPTURE_8:
substream = chip->capture_substream;
if (chip->capture_format == SB_DSP_INPUT)
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 5363d88cc4b9..2e5a5c5279e8 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu)
}
/*
- * This is apparently the standard way to initailise an MPU-401
+ * This is apparently the standard way to initialise an MPU-401
*/
static inline void initialise_mpu401(const struct snd_mpu401 *mpu)
{
@@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c)
}
/*
- * Tell the SoundScape to begin a DMA tranfer using the given channel.
+ * Tell the SoundScape to begin a DMA transfer using the given channel.
* All locking issues are left to the caller.
*/
static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg)
@@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu)
}
/*
- * Initialse an MPU-401 subdevice for MIDI support on the SoundScape.
+ * Initialise an MPU-401 subdevice for MIDI support on the SoundScape.
*/
static int create_mpu401(struct snd_card *card, int devnum,
unsigned long port, int irq)
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 823ccfa089b2..81c6a9830727 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg)
tt_dmasnd.input_gain =
RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 |
RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff);
- /* fall through - return set value */
+ fallthrough; /* return set value */
case SOUND_MIXER_READ_MIC:
return IOCTL_OUT(arg,
RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) |
diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c
index f802ea331e24..38f25e97538f 100644
--- a/sound/oss/dmasound/dmasound_core.c
+++ b/sound/oss/dmasound/dmasound_core.c
@@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str)
printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius);
else
catchRadius = ints[3];
- /* fall through */
+ fallthrough;
case 2:
if (ints[1] < MIN_BUFFERS)
printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs);
else
numWriteBufs = ints[1];
- /* fall through */
+ fallthrough;
case 1:
if ((size = ints[2]) < 256) /* check for small buffer specs */
size <<= 10 ;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 6758c072000e..012a7ee849e8 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg)
case AC97_ID_ST_AC97_ID4:
if (reg == 0x08)
return 0;
- /* fall through */
+ fallthrough;
case AC97_ID_ST7597:
if (reg == 0x22 || reg == 0x7a)
return 1;
- /* fall through */
+ fallthrough;
case AC97_ID_AK4540:
case AC97_ID_AK4542:
if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c)
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 45ef0f52ec55..1627a74baf3c 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -19,7 +19,7 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
const char *name);
static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
const unsigned int *tlv,
- const char * const *slaves);
+ const char * const *followers);
/*
* Chip specific initialization
@@ -1791,10 +1791,10 @@ static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = {
AC97_SINGLE("Line Jack Sense", AC97_AD_JACK_SPDIF, 12, 1, 0),
};
-/* black list to avoid HP/Line jack-sense controls
+/* deny list to avoid HP/Line jack-sense controls
* (SS vendor << 16 | device)
*/
-static const unsigned int ad1981_jacks_blacklist[] = {
+static const unsigned int ad1981_jacks_denylist[] = {
0x10140523, /* Thinkpad R40 */
0x10140534, /* Thinkpad X31 */
0x10140537, /* Thinkpad T41p */
@@ -1821,7 +1821,7 @@ static int check_list(struct snd_ac97 *ac97, const unsigned int *list)
static int patch_ad1981a_specific(struct snd_ac97 * ac97)
{
- if (check_list(ac97, ad1981_jacks_blacklist))
+ if (check_list(ac97, ad1981_jacks_denylist))
return 0;
return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
@@ -1835,10 +1835,10 @@ static const struct snd_ac97_build_ops patch_ad1981a_build_ops = {
#endif
};
-/* white list to enable HP jack-sense bits
+/* allow list to enable HP jack-sense bits
* (SS vendor << 16 | device)
*/
-static const unsigned int ad1981_jacks_whitelist[] = {
+static const unsigned int ad1981_jacks_allowlist[] = {
0x0e11005a, /* HP nc4000/4010 */
0x103c0890, /* HP nc6000 */
0x103c0938, /* HP nc4220 */
@@ -1853,7 +1853,7 @@ static const unsigned int ad1981_jacks_whitelist[] = {
static void check_ad1981_hp_jack_sense(struct snd_ac97 *ac97)
{
- if (check_list(ac97, ad1981_jacks_whitelist))
+ if (check_list(ac97, ad1981_jacks_allowlist))
/* enable headphone jack sense */
snd_ac97_update_bits(ac97, AC97_AD_JACK_SPDIF, 1<<11, 1<<11);
}
@@ -1877,7 +1877,7 @@ static int patch_ad1981b_specific(struct snd_ac97 *ac97)
if ((err = patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1)) < 0)
return err;
- if (check_list(ac97, ad1981_jacks_blacklist))
+ if (check_list(ac97, ad1981_jacks_denylist))
return 0;
return patch_build_controls(ac97, snd_ac97_ad1981x_jack_sense,
ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
@@ -3373,7 +3373,7 @@ AC97_SINGLE("Downmix LFE and Center to Front", 0x5a, 12, 1, 0),
AC97_SINGLE("Downmix Surround to Front", 0x5a, 11, 1, 0),
};
-static const char * const slave_vols_vt1616[] = {
+static const char * const follower_vols_vt1616[] = {
"Front Playback Volume",
"Surround Playback Volume",
"Center Playback Volume",
@@ -3381,7 +3381,7 @@ static const char * const slave_vols_vt1616[] = {
NULL
};
-static const char * const slave_sws_vt1616[] = {
+static const char * const follower_sws_vt1616[] = {
"Front Playback Switch",
"Surround Playback Switch",
"Center Playback Switch",
@@ -3400,10 +3400,10 @@ static struct snd_kcontrol *snd_ac97_find_mixer_ctl(struct snd_ac97 *ac97,
return snd_ctl_find_id(ac97->bus->card, &id);
}
-/* create a virtual master control and add slaves */
+/* create a virtual master control and add followers */
static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
const unsigned int *tlv,
- const char * const *slaves)
+ const char * const *followers)
{
struct snd_kcontrol *kctl;
const char * const *s;
@@ -3416,16 +3416,16 @@ static int snd_ac97_add_vmaster(struct snd_ac97 *ac97, char *name,
if (err < 0)
return err;
- for (s = slaves; *s; s++) {
+ for (s = followers; *s; s++) {
struct snd_kcontrol *sctl;
sctl = snd_ac97_find_mixer_ctl(ac97, *s);
if (!sctl) {
dev_dbg(ac97->bus->card->dev,
- "Cannot find slave %s, skipped\n", *s);
+ "Cannot find follower %s, skipped\n", *s);
continue;
}
- err = snd_ctl_add_slave(kctl, sctl);
+ err = snd_ctl_add_follower(kctl, sctl);
if (err < 0)
return err;
}
@@ -3451,12 +3451,12 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97)
snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Front Playback");
err = snd_ac97_add_vmaster(ac97, "Master Playback Volume",
- kctl->tlv.p, slave_vols_vt1616);
+ kctl->tlv.p, follower_vols_vt1616);
if (err < 0)
return err;
err = snd_ac97_add_vmaster(ac97, "Master Playback Switch",
- NULL, slave_sws_vt1616);
+ NULL, follower_sws_vt1616);
if (err < 0)
return err;
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index a9540c2c4a1a..023c35a2a951 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -1904,7 +1904,7 @@ static int snd_asihpi_tuner_band_get(struct snd_kcontrol *kcontrol,
*/
u16 band, idx;
u16 tuner_bands[HPI_TUNER_BAND_LAST];
- u32 num_bands = 0;
+ __always_unused u32 num_bands;
num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands,
HPI_TUNER_BAND_LAST);
@@ -1931,7 +1931,7 @@ static int snd_asihpi_tuner_band_put(struct snd_kcontrol *kcontrol,
unsigned int idx;
u16 band;
u16 tuner_bands[HPI_TUNER_BAND_LAST];
- u32 num_bands = 0;
+ __always_unused u32 num_bands;
num_bands = asihpi_tuner_band_query(kcontrol, tuner_bands,
HPI_TUNER_BAND_LAST);
@@ -2161,7 +2161,6 @@ static int snd_card_asihpi_mux_count_sources(struct snd_kcontrol *snd_control)
static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- int err;
u16 src_node_type, src_node_index;
u32 h_control = kcontrol->private_value;
@@ -2174,10 +2173,9 @@ static int snd_asihpi_mux_info(struct snd_kcontrol *kcontrol,
uinfo->value.enumerated.item =
uinfo->value.enumerated.items - 1;
- err =
- hpi_multiplexer_query_source(h_control,
- uinfo->value.enumerated.item,
- &src_node_type, &src_node_index);
+ hpi_multiplexer_query_source(h_control,
+ uinfo->value.enumerated.item,
+ &src_node_type, &src_node_index);
sprintf(uinfo->value.enumerated.name, "%s %d",
asihpi_src_names[src_node_type - HPI_SOURCENODE_NONE],
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index ad912f9dac7e..6859d51389f5 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -53,7 +53,7 @@ If handle is invalid *pPhysicalAddr is set to zero and return 1
u16 hpios_locked_mem_get_phys_addr(struct consistent_dma_area
*locked_mem_handle, u32 *p_physical_addr);
-/** Get the CPU address of of memory represented by LockedMemHandle.
+/** Get the CPU address of memory represented by LockedMemHandle.
If handle is NULL *ppvVirtualAddr is set to NULL and return 1
*/
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 968510bc2552..7d1abaedb46a 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -28,10 +28,12 @@ struct hpi_adapters_list {
static struct hpi_adapters_list adapters;
/**
-* Given an HPI Message that was sent out and a response that was received,
-* validate that the response has the correct fields filled in,
-* i.e ObjectType, Function etc
-**/
+ * hpi_validate_response - Given an HPI Message that was sent out and
+ * a response that was received, validate that the response has the
+ * correct fields filled in, i.e ObjectType, Function etc
+ * @phm: message
+ * @phr: response
+ */
u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr)
{
if (phr->type != HPI_TYPE_RESPONSE) {
@@ -106,10 +108,11 @@ void hpi_delete_adapter(struct hpi_adapter_obj *pao)
}
/**
-* FindAdapter returns a pointer to the struct hpi_adapter_obj with
-* index wAdapterIndex in an HPI_ADAPTERS_LIST structure.
-*
-*/
+ * hpi_find_adapter - FindAdapter returns a pointer to the struct
+ * hpi_adapter_obj with index wAdapterIndex in an HPI_ADAPTERS_LIST
+ * structure.
+ * @adapter_index: value in [0, HPI_MAX_ADAPTERS[
+ */
struct hpi_adapter_obj *hpi_find_adapter(u16 adapter_index)
{
struct hpi_adapter_obj *pao = NULL;
@@ -137,10 +140,9 @@ struct hpi_adapter_obj *hpi_find_adapter(u16 adapter_index)
}
/**
-*
-* wipe an HPI_ADAPTERS_LIST structure.
-*
-**/
+ * wipe_adapter_list - wipe an HPI_ADAPTERS_LIST structure.
+ *
+ */
static void wipe_adapter_list(void)
{
memset(&adapters, 0, sizeof(adapters));
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 85d3b4e95489..a25d75455802 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream)
case 8:
data |= ATI_REG_OUT_DMA_SLOT_BIT(10) |
ATI_REG_OUT_DMA_SLOT_BIT(11);
- /* fall through */
+ fallthrough;
case 6:
data |= ATI_REG_OUT_DMA_SLOT_BIT(7) |
ATI_REG_OUT_DMA_SLOT_BIT(8);
- /* fall through */
+ fallthrough;
case 4:
data |= ATI_REG_OUT_DMA_SLOT_BIT(6) |
ATI_REG_OUT_DMA_SLOT_BIT(9);
- /* fall through */
+ fallthrough;
default:
data |= ATI_REG_OUT_DMA_SLOT_BIT(3) |
ATI_REG_OUT_DMA_SLOT_BIT(4);
diff --git a/sound/pci/au88x0/au88x0_a3ddata.c b/sound/pci/au88x0/au88x0_a3ddata.c
index 18623cb6bc52..a5da3b3a546a 100644
--- a/sound/pci/au88x0/au88x0_a3ddata.c
+++ b/sound/pci/au88x0/au88x0_a3ddata.c
@@ -21,7 +21,7 @@ static const a3d_Hrtf_t A3dHrirZeros = {
0, 0, 0
};
-static const a3d_Hrtf_t A3dHrirImpulse = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirImpulse = {
0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0,
0, 0, 0, 0,
@@ -30,7 +30,7 @@ static const a3d_Hrtf_t A3dHrirImpulse = {
0, 0, 0
};
-static const a3d_Hrtf_t A3dHrirOnes = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirOnes = {
0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
0x7fff,
0x7fff,
@@ -47,7 +47,7 @@ static const a3d_Hrtf_t A3dHrirOnes = {
0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff
};
-static const a3d_Hrtf_t A3dHrirSatTest = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirSatTest = {
0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff, 0x7fff,
0x7fff,
0x7fff,
@@ -59,7 +59,7 @@ static const a3d_Hrtf_t A3dHrirSatTest = {
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
};
-static const a3d_Hrtf_t A3dHrirDImpulse = {
+static __maybe_unused const a3d_Hrtf_t A3dHrirDImpulse = {
0, 0x7fff, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0,
0, 0, 0, 0,
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index f5512b72b3e0..5180f1bd1326 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1103,7 +1103,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0xc,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
- /* fall through */
+ fallthrough;
/* 3 pages */
case 3:
dma->cfg0 |= 0x12000000;
@@ -1111,14 +1111,14 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x8,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
- /* fall through */
+ fallthrough;
/* 2 pages */
case 2:
dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize - 1);
hwwrite(vortex->mmio,
VORTEX_ADBDMA_BUFBASE + (adbdma << 4) + 0x4,
snd_pcm_sgbuf_get_addr(dma->substream, psize));
- /* fall through */
+ fallthrough;
/* 1 page */
case 1:
dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize - 1) << 0xc);
@@ -1381,20 +1381,20 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma,
dma->cfg1 |= 0x88000000 | 0x44000000 | 0x30000000 | (psize-1);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0xc,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 3));
- /* fall through */
+ fallthrough;
/* 3 pages */
case 3:
dma->cfg0 |= 0x12000000;
dma->cfg1 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x8,
snd_pcm_sgbuf_get_addr(dma->substream, psize * 2));
- /* fall through */
+ fallthrough;
/* 2 pages */
case 2:
dma->cfg0 |= 0x88000000 | 0x44000000 | 0x10000000 | (psize-1);
hwwrite(vortex->mmio, VORTEX_WTDMA_BUFBASE + (wtdma << 4) + 0x4,
snd_pcm_sgbuf_get_addr(dma->substream, psize));
- /* fall through */
+ fallthrough;
/* 1 page */
case 1:
dma->cfg0 |= 0x80000000 | 0x40000000 | ((psize-1) << 0xc);
diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c
index 084fcbf8ae80..27859536d7c0 100644
--- a/sound/pci/au88x0/au88x0_xtalk.c
+++ b/sound/pci/au88x0/au88x0_xtalk.c
@@ -17,35 +17,35 @@
static short const sXtalkWideKLeftEq = 0x269C;
static short const sXtalkWideKRightEq = 0x269C;
static short const sXtalkWideKLeftXt = 0xF25E;
-static short const sXtalkWideKRightXt = 0xF25E;
+static __maybe_unused short const sXtalkWideKRightXt = 0xF25E;
static short const sXtalkWideShiftLeftEq = 1;
static short const sXtalkWideShiftRightEq = 1;
static short const sXtalkWideShiftLeftXt = 0;
-static short const sXtalkWideShiftRightXt = 0;
+static __maybe_unused short const sXtalkWideShiftRightXt = 0;
static unsigned short const wXtalkWideLeftDelay = 0xd;
static unsigned short const wXtalkWideRightDelay = 0xd;
static short const sXtalkNarrowKLeftEq = 0x468D;
static short const sXtalkNarrowKRightEq = 0x468D;
static short const sXtalkNarrowKLeftXt = 0xF82E;
-static short const sXtalkNarrowKRightXt = 0xF82E;
+static __maybe_unused short const sXtalkNarrowKRightXt = 0xF82E;
static short const sXtalkNarrowShiftLeftEq = 0x3;
static short const sXtalkNarrowShiftRightEq = 0x3;
static short const sXtalkNarrowShiftLeftXt = 0;
-static short const sXtalkNarrowShiftRightXt = 0;
+static __maybe_unused short const sXtalkNarrowShiftRightXt = 0;
static unsigned short const wXtalkNarrowLeftDelay = 0x7;
static unsigned short const wXtalkNarrowRightDelay = 0x7;
-static xtalk_gains_t const asXtalkGainsDefault = {
+static __maybe_unused xtalk_gains_t const asXtalkGainsDefault = {
0x4000, 0x4000, 0x4000, 0x4000, 0x4000,
0x4000, 0x4000, 0x4000, 0x4000, 0x4000
};
-static xtalk_gains_t const asXtalkGainsTest = {
+static __maybe_unused xtalk_gains_t const asXtalkGainsTest = {
0x7fff, 0x8000, 0x0000, 0x0000, 0x0001,
0xffff, 0x4000, 0xc000, 0x0002, 0xfffe
};
-static xtalk_gains_t const asXtalkGains1Chan = {
+static __maybe_unused xtalk_gains_t const asXtalkGains1Chan = {
0x7FFF, 0, 0, 0, 0,
0x7FFF, 0, 0, 0, 0,
};
@@ -64,7 +64,7 @@ static xtalk_dline_t const alXtalkDlineZeros = {
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
};
-static xtalk_dline_t const alXtalkDlineTest = {
+static __maybe_unused xtalk_dline_t const alXtalkDlineTest = {
0x0000fc18, 0xfff03e8, 0x000186a0, 0xfffe7960, 1, 0xffffffff, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0
@@ -74,7 +74,7 @@ static xtalk_instate_t const asXtalkInStateZeros = {
0, 0, 0, 0
};
-static xtalk_instate_t const asXtalkInStateTest = {
+static __maybe_unused xtalk_instate_t const asXtalkInStateTest = {
0x0080, 0xff80, 0x0001, 0xffff
};
@@ -89,11 +89,11 @@ static xtalk_state_t const asXtalkOutStateZeros = {
static short const sDiamondKLeftEq = 0x401d;
static short const sDiamondKRightEq = 0x401d;
static short const sDiamondKLeftXt = 0xF90E;
-static short const sDiamondKRightXt = 0xF90E;
+static __maybe_unused short const sDiamondKRightXt = 0xF90E;
static short const sDiamondShiftLeftEq = 1;
static short const sDiamondShiftRightEq = 1;
static short const sDiamondShiftLeftXt = 0;
-static short const sDiamondShiftRightXt = 0;
+static __maybe_unused short const sDiamondShiftRightXt = 0;
static unsigned short const wDiamondLeftDelay = 0xb;
static unsigned short const wDiamondRightDelay = 0xb;
@@ -118,7 +118,7 @@ static xtalk_coefs_t const asXtalkWideCoefsLeftXt = {
{0x77dc, 0xc79e, 0xffb8, 0x000a, 0},
{0, 0, 0, 0, 0}
};
-static xtalk_coefs_t const asXtalkWideCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asXtalkWideCoefsRightXt = {
{0x55c6, 0xc97b, 0x005b, 0x0047, 0},
{0x6a60, 0xca20, 0xffc6, 0x0040, 0},
{0x6411, 0xd711, 0xfca1, 0x0190, 0},
@@ -149,7 +149,7 @@ static xtalk_coefs_t const asXtalkNarrowCoefsLeftXt = {
{0, 0, 0, 0, 0}
};
-static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
{0x3CB2, 0xDF49, 0xF6EA, 0x095B, 0},
{0x6777, 0xC915, 0xFEAF, 0x00B1, 0},
{0x7762, 0xC7D9, 0x025B, 0xFDA6, 0},
@@ -172,7 +172,7 @@ static xtalk_coefs_t const asXtalkCoefsPipe = {
{0, 0, 0x0FA0, 0, 0},
{0, 0, 0x1180, 0, 0},
};
-static xtalk_coefs_t const asXtalkCoefsNegPipe = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsNegPipe = {
{0, 0, 0xF380, 0, 0},
{0, 0, 0xF380, 0, 0},
{0, 0, 0xF380, 0, 0},
@@ -180,7 +180,7 @@ static xtalk_coefs_t const asXtalkCoefsNegPipe = {
{0, 0, 0xF200, 0, 0}
};
-static xtalk_coefs_t const asXtalkCoefsNumTest = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsNumTest = {
{0, 0, 0xF380, 0x8000, 0x6D60},
{0, 0, 0, 0, 0},
{0, 0, 0, 0, 0},
@@ -188,7 +188,7 @@ static xtalk_coefs_t const asXtalkCoefsNumTest = {
{0, 0, 0, 0, 0}
};
-static xtalk_coefs_t const asXtalkCoefsDenTest = {
+static __maybe_unused xtalk_coefs_t const asXtalkCoefsDenTest = {
{0xC000, 0x2000, 0x4000, 0, 0},
{0, 0, 0, 0, 0},
{0, 0, 0, 0, 0},
@@ -196,7 +196,7 @@ static xtalk_coefs_t const asXtalkCoefsDenTest = {
{0, 0, 0, 0, 0}
};
-static xtalk_state_t const asXtalkOutStateTest = {
+static __maybe_unused xtalk_state_t const asXtalkOutStateTest = {
{0x7FFF, 0x0004, 0xFFFC, 0},
{0xFE00, 0x0008, 0xFFF8, 0x4000},
{0x0200, 0x0010, 0xFFF0, 0xC000},
@@ -228,7 +228,7 @@ static xtalk_coefs_t const asDiamondCoefsLeftXt = {
{0, 0, 0, 0, 0}
};
-static xtalk_coefs_t const asDiamondCoefsRightXt = {
+static __maybe_unused xtalk_coefs_t const asDiamondCoefsRightXt = {
{0x3B50, 0xFE08, 0xF959, 0x0060, 0},
{0x9FCB, 0xD8F1, 0x00A2, 0x003A, 0},
{0, 0, 0, 0, 0},
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
index 4e64eb5d8f64..c84f1a45194f 100644
--- a/sound/pci/aw2/aw2-saa7146.c
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -330,7 +330,7 @@ void snd_aw2_saa7146_pcm_trigger_stop_capture(struct snd_aw2_saa7146 *chip,
irqreturn_t snd_aw2_saa7146_interrupt(int irq, void *dev_id)
{
unsigned int isr;
- unsigned int iicsta;
+ __always_unused unsigned int iicsta;
struct snd_aw2_saa7146 *chip = dev_id;
isr = READREG(ISR);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 58167d8469e1..77c7030ebbfa 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
default:
snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
- /* fall-through */
+ fallthrough;
case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 6567504665b9..54cb223caa2f 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -30,7 +30,7 @@ static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */
static int digital_rate[SNDRV_CARDS]; /* digital input rate */
-static bool load_all; /* allow to load the non-whitelisted cards */
+static bool load_all; /* allow to load cards not the allowlist */
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for Bt87x soundcard");
@@ -41,7 +41,7 @@ MODULE_PARM_DESC(enable, "Enable Bt87x soundcard");
module_param_array(digital_rate, int, NULL, 0444);
MODULE_PARM_DESC(digital_rate, "Digital input rate for Bt87x soundcard");
module_param(load_all, bool, 0444);
-MODULE_PARM_DESC(load_all, "Allow to load the non-whitelisted cards");
+MODULE_PARM_DESC(load_all, "Allow to load cards not on the allowlist");
/* register offsets */
@@ -801,7 +801,7 @@ MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
* (DVB cards use the audio function to transfer MPEG data) */
static struct {
unsigned short subvendor, subdevice;
-} blacklist[] = {
+} denylist[] = {
{0x0071, 0x0101}, /* Nebula Electronics DigiTV */
{0x11bd, 0x001c}, /* Pinnacle PCTV Sat */
{0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */
@@ -817,7 +817,7 @@ static struct {
static struct pci_driver driver;
-/* return the id of the card, or a negative value if it's blacklisted */
+/* return the id of the card, or a negative value if it's on the denylist */
static int snd_bt87x_detect_card(struct pci_dev *pci)
{
int i;
@@ -827,9 +827,9 @@ static int snd_bt87x_detect_card(struct pci_dev *pci)
if (supported && supported->driver_data > 0)
return supported->driver_data;
- for (i = 0; i < ARRAY_SIZE(blacklist); ++i)
- if (blacklist[i].subvendor == pci->subsystem_vendor &&
- blacklist[i].subdevice == pci->subsystem_device) {
+ for (i = 0; i < ARRAY_SIZE(denylist); ++i)
+ if (denylist[i].subvendor == pci->subsystem_vendor &&
+ denylist[i].subdevice == pci->subsystem_device) {
dev_dbg(&pci->dev,
"card %#04x-%#04x:%#04x has no audio\n",
pci->device, pci->subsystem_vendor, pci->subsystem_device);
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 3b8ec673dc0a..c852c6a75b91 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -739,7 +739,7 @@ static int rename_ctl(struct snd_card *card, const char *src, const char *dst)
static
DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
"Analog Front Playback Volume",
"Analog Rear Playback Volume",
"Analog Center/LFE Playback Volume",
@@ -752,7 +752,7 @@ static const char * const slave_vols[] = {
NULL
};
-static const char * const slave_sws[] = {
+static const char * const follower_sws[] = {
"Analog Front Playback Switch",
"Analog Rear Playback Switch",
"Analog Center/LFE Playback Switch",
@@ -761,13 +761,13 @@ static const char * const slave_sws[] = {
NULL
};
-static void add_slaves(struct snd_card *card,
- struct snd_kcontrol *master, const char * const *list)
+static void add_followers(struct snd_card *card,
+ struct snd_kcontrol *master, const char * const *list)
{
for (; *list; list++) {
- struct snd_kcontrol *slave = ctl_find(card, *list);
- if (slave)
- snd_ctl_add_slave(master, slave);
+ struct snd_kcontrol *follower = ctl_find(card, *list);
+ if (follower)
+ snd_ctl_add_follower(master, follower);
}
}
@@ -852,7 +852,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu)
err = snd_ctl_add(card, vmaster);
if (err < 0)
return err;
- add_slaves(card, vmaster, slave_vols);
+ add_followers(card, vmaster, follower_vols);
if (emu->details->spi_dac) {
vmaster = snd_ctl_make_virtual_master("Master Playback Switch",
@@ -862,7 +862,7 @@ int snd_ca0106_mixer(struct snd_ca0106 *emu)
err = snd_ctl_add(card, vmaster);
if (err < 0)
return err;
- add_slaves(card, vmaster, slave_sws);
+ add_followers(card, vmaster, follower_sws);
}
strcpy(card->mixername, "CA0106");
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index a080d63a9b45..4490dd7469d9 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -766,7 +766,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned
rate = 48000 / 9;
/*
- * We can not capture at at rate greater than the Input Rate (48000).
+ * We can not capture at a rate greater than the Input Rate (48000).
* Return an error if an attempt is made to stray outside that limit.
*/
if (rate > 48000)
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 6b536fc23ca6..1f90ca723f4d 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1716,7 +1716,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip)
struct dsp_spos_instance * ins = chip->dsp_spos_instance;
if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
- /* remove AsynchFGTxSCB and and PCMSerialInput_II */
+ /* remove AsynchFGTxSCB and PCMSerialInput_II */
cs46xx_dsp_disable_spdif_out (chip);
/* save state */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index e56a230f6a9c..f8ac96cf38a4 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1282,7 +1282,7 @@ static int atc_identify_card(struct ct_atc *atc, unsigned int ssid)
if (p) {
if (p->value < 0) {
dev_err(atc->card->dev,
- "Device %04x:%04x is black-listed\n",
+ "Device %04x:%04x is on the denylist\n",
vendor_id, device_id);
return -ENOENT;
}
@@ -1655,6 +1655,10 @@ static const struct ct_atc atc_preset = {
* ct_atc_create - create and initialize a hardware manager
* @card: corresponding alsa card object
* @pci: corresponding kernel pci device object
+ * @rsr: reference sampling rate
+ * @msr: master sampling rate
+ * @chip_type: CHIPTYP enum values
+ * @ssid: vendor ID (upper 16 bits) and device ID (lower 16 bits)
* @ratc: return created object address in it
*
* Creates and initializes a hardware manager.
diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c
index 9b7e63f4a3a7..1d5064486217 100644
--- a/sound/pci/ctxfi/cthardware.c
+++ b/sound/pci/ctxfi/cthardware.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File cthardware.c
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index 015c0d676897..108ab449c968 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File cthw20k1.c
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index ce44cbe6459f..fc1bc18caee9 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File cthw20k2.c
diff --git a/sound/pci/ctxfi/ctimap.c b/sound/pci/ctxfi/ctimap.c
index eb1825e13fc5..d5a53d2f5f15 100644
--- a/sound/pci/ctxfi/ctimap.c
+++ b/sound/pci/ctxfi/ctimap.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File ctimap.c
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
index 84514dc90d87..6797fde3d788 100644
--- a/sound/pci/ctxfi/ctmixer.c
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File ctmixer.c
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
index 6ee6a9675ca5..3f48ad0e27e7 100644
--- a/sound/pci/ctxfi/ctpcm.c
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -1,5 +1,5 @@
// SPDX-License-Identifier: GPL-2.0-only
-/**
+/*
* Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
*
* @File ctpcm.c
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 0941a7a17623..a20b2bb5c898 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2,6 +2,7 @@
/*
* ALSA driver for Echoaudio soundcards.
* Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ * Copyright (C) 2020 Mark Hills <mark@xwax.org>
*/
#include <linux/module.h>
@@ -245,13 +246,20 @@ static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
SNDRV_PCM_HW_PARAM_RATE);
struct echoaudio *chip = rule->private;
struct snd_interval fixed;
+ int err;
+
+ mutex_lock(&chip->mode_mutex);
- if (!chip->can_set_rate) {
+ if (chip->can_set_rate) {
+ err = 0;
+ } else {
snd_interval_any(&fixed);
fixed.min = fixed.max = chip->sample_rate;
- return snd_interval_refine(rate, &fixed);
+ err = snd_interval_refine(rate, &fixed);
}
- return 0;
+
+ mutex_unlock(&chip->mode_mutex);
+ return err;
}
@@ -322,7 +330,7 @@ static int pcm_open(struct snd_pcm_substream *substream,
SNDRV_PCM_HW_PARAM_RATE, -1)) < 0)
return err;
- /* Finally allocate a page for the scatter-gather list */
+ /* Allocate a page for the scatter-gather list */
if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
&chip->pci->dev,
PAGE_SIZE, &pipe->sgpage)) < 0) {
@@ -330,6 +338,17 @@ static int pcm_open(struct snd_pcm_substream *substream,
return err;
}
+ /*
+ * Sole ownership required to set the rate
+ */
+
+ dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
+ chip->opencount, chip->can_set_rate, chip->rate_set);
+
+ chip->opencount++;
+ if (chip->opencount > 1 && chip->rate_set)
+ chip->can_set_rate = 0;
+
return 0;
}
@@ -353,12 +372,7 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream)
hw_rule_capture_format_by_channels, NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
return err;
- atomic_inc(&chip->opencount);
- if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
- chip->can_set_rate=0;
- dev_dbg(chip->card->dev, "pcm_analog_in_open cs=%d oc=%d r=%d\n",
- chip->can_set_rate, atomic_read(&chip->opencount),
- chip->sample_rate);
+
return 0;
}
@@ -388,12 +402,7 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream)
NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
return err;
- atomic_inc(&chip->opencount);
- if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
- chip->can_set_rate=0;
- dev_dbg(chip->card->dev, "pcm_analog_out_open cs=%d oc=%d r=%d\n",
- chip->can_set_rate, atomic_read(&chip->opencount),
- chip->sample_rate);
+
return 0;
}
@@ -429,10 +438,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
goto din_exit;
- atomic_inc(&chip->opencount);
- if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
- chip->can_set_rate=0;
-
din_exit:
mutex_unlock(&chip->mode_mutex);
return err;
@@ -471,9 +476,7 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream)
NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
-1)) < 0)
goto dout_exit;
- atomic_inc(&chip->opencount);
- if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
- chip->can_set_rate=0;
+
dout_exit:
mutex_unlock(&chip->mode_mutex);
return err;
@@ -488,23 +491,29 @@ dout_exit:
static int pcm_close(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
- int oc;
/* Nothing to do here. Audio is already off and pipe will be
* freed by its callback
*/
- atomic_dec(&chip->opencount);
- oc = atomic_read(&chip->opencount);
- dev_dbg(chip->card->dev, "pcm_close oc=%d cs=%d rs=%d\n", oc,
- chip->can_set_rate, chip->rate_set);
- if (oc < 2)
+ mutex_lock(&chip->mode_mutex);
+
+ dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
+ chip->opencount, chip->can_set_rate, chip->rate_set);
+
+ chip->opencount--;
+
+ switch (chip->opencount) {
+ case 1:
chip->can_set_rate = 1;
- if (oc == 0)
+ break;
+
+ case 0:
chip->rate_set = 0;
- dev_dbg(chip->card->dev, "pcm_close2 oc=%d cs=%d rs=%d\n", oc,
- chip->can_set_rate, chip->rate_set);
+ break;
+ }
+ mutex_unlock(&chip->mode_mutex);
return 0;
}
@@ -582,7 +591,7 @@ static int init_engine(struct snd_pcm_substream *substream,
/* This stuff is used by the irq handler, so it must be
* initialized before chip->substream
*/
- chip->last_period[pipe_index] = 0;
+ pipe->last_period = 0;
pipe->last_counter = 0;
pipe->position = 0;
smp_wmb();
@@ -690,7 +699,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
break;
case SNDRV_PCM_FORMAT_S32_BE:
format.data_are_bigendian = 1;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_FORMAT_S32_LE:
format.bits_per_sample = 32;
break;
@@ -703,9 +712,22 @@ static int pcm_prepare(struct snd_pcm_substream *substream)
if (snd_BUG_ON(pipe_index >= px_num(chip)))
return -EINVAL;
- if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index)))
+
+ /*
+ * We passed checks we can do independently; now take
+ * exclusive control
+ */
+
+ spin_lock_irq(&chip->lock);
+
+ if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index))) {
+ spin_unlock_irq(&chip->lock);
return -EINVAL;
+ }
+
set_audio_format(chip, pipe_index, &format);
+ spin_unlock_irq(&chip->lock);
+
return 0;
}
@@ -738,11 +760,11 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
pipe = chip->substream[i]->runtime->private_data;
switch (pipe->state) {
case PIPE_STATE_STOPPED:
- chip->last_period[i] = 0;
+ pipe->last_period = 0;
pipe->last_counter = 0;
pipe->position = 0;
*pipe->dma_counter = 0;
- /* fall through */
+ fallthrough;
case PIPE_STATE_PAUSED:
pipe->state = PIPE_STATE_STARTED;
break;
@@ -786,19 +808,26 @@ static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct audiopipe *pipe = runtime->private_data;
- size_t cnt, bufsize, pos;
+ u32 counter, step;
- cnt = le32_to_cpu(*pipe->dma_counter);
- pipe->position += cnt - pipe->last_counter;
- pipe->last_counter = cnt;
- bufsize = substream->runtime->buffer_size;
- pos = bytes_to_frames(substream->runtime, pipe->position);
+ /*
+ * IRQ handling runs concurrently. Do not share tracking of
+ * counter with it, which would race or require locking
+ */
- while (pos >= bufsize) {
- pipe->position -= frames_to_bytes(substream->runtime, bufsize);
- pos -= bufsize;
- }
- return pos;
+ counter = le32_to_cpu(*pipe->dma_counter); /* presumed atomic */
+
+ step = counter - pipe->last_counter; /* handles wrapping */
+ pipe->last_counter = counter;
+
+ /* counter doesn't neccessarily wrap on a multiple of
+ * buffer_size, so can't derive the position; must
+ * accumulate */
+
+ pipe->position += step;
+ pipe->position %= frames_to_bytes(runtime, runtime->buffer_size); /* wrap */
+
+ return bytes_to_frames(runtime, pipe->position);
}
@@ -1409,7 +1438,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
/* Do not allow the user to change the digital mode when a pcm
device is open because it also changes the number of channels
and the allowed sample rates */
- if (atomic_read(&chip->opencount)) {
+ if (chip->opencount) {
changed = -EAGAIN;
} else {
changed = set_digital_mode(chip, dmode);
@@ -1761,14 +1790,43 @@ static const struct snd_kcontrol_new snd_echo_channels_info = {
/******************************************************************************
- IRQ Handler
+ IRQ Handling
******************************************************************************/
+/* Check if a period has elapsed since last interrupt
+ *
+ * Don't make any updates to state; PCM core handles this with the
+ * correct locks.
+ *
+ * \return true if a period has elapsed, otherwise false
+ */
+static bool period_has_elapsed(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audiopipe *pipe = runtime->private_data;
+ u32 counter, step;
+ size_t period_bytes;
+
+ if (pipe->state != PIPE_STATE_STARTED)
+ return false;
+
+ period_bytes = frames_to_bytes(runtime, runtime->period_size);
+
+ counter = le32_to_cpu(*pipe->dma_counter); /* presumed atomic */
+
+ step = counter - pipe->last_period; /* handles wrapping */
+ step -= step % period_bytes; /* acknowledge whole periods only */
+
+ if (step == 0)
+ return false; /* haven't advanced a whole period yet */
+
+ pipe->last_period += step; /* used exclusively by us */
+ return true;
+}
static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
{
struct echoaudio *chip = dev_id;
- struct snd_pcm_substream *substream;
- int period, ss, st;
+ int ss, st;
spin_lock(&chip->lock);
st = service_irq(chip);
@@ -1779,17 +1837,13 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
/* The hardware doesn't tell us which substream caused the irq,
thus we have to check all running substreams. */
for (ss = 0; ss < DSP_MAXPIPES; ss++) {
+ struct snd_pcm_substream *substream;
+
substream = chip->substream[ss];
- if (substream && ((struct audiopipe *)substream->runtime->
- private_data)->state == PIPE_STATE_STARTED) {
- period = pcm_pointer(substream) /
- substream->runtime->period_size;
- if (period != chip->last_period[ss]) {
- chip->last_period[ss] = period;
- spin_unlock(&chip->lock);
- snd_pcm_period_elapsed(substream);
- spin_lock(&chip->lock);
- }
+ if (substream && period_has_elapsed(substream)) {
+ spin_unlock(&chip->lock);
+ snd_pcm_period_elapsed(substream);
+ spin_lock(&chip->lock);
}
}
spin_unlock(&chip->lock);
@@ -1874,7 +1928,7 @@ static int snd_echo_create(struct snd_card *card,
chip->card = card;
chip->pci = pci;
chip->irq = -1;
- atomic_set(&chip->opencount, 0);
+ chip->opencount = 0;
mutex_init(&chip->mode_mutex);
chip->can_set_rate = 1;
} else {
@@ -1896,8 +1950,7 @@ static int snd_echo_create(struct snd_card *card,
snd_echo_free(chip);
return -EBUSY;
}
- chip->dsp_registers = (volatile u32 __iomem *)
- ioremap(chip->dsp_registers_phys, sz);
+ chip->dsp_registers = ioremap(chip->dsp_registers_phys, sz);
if (!chip->dsp_registers) {
dev_err(chip->card->dev, "ioremap failed\n");
snd_echo_free(chip);
@@ -1955,7 +2008,8 @@ static int snd_echo_probe(struct pci_dev *pci,
struct snd_card *card;
struct echoaudio *chip;
char *dsp;
- int i, err;
+ __maybe_unused int i;
+ int err;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@@ -2158,7 +2212,6 @@ static int snd_echo_resume(struct device *dev)
if (err < 0) {
kfree(commpage_bak);
dev_err(dev, "resume init_hw err=%d\n", err);
- snd_echo_free(chip);
return err;
}
@@ -2185,7 +2238,6 @@ static int snd_echo_resume(struct device *dev)
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
KBUILD_MODNAME, chip)) {
dev_err(chip->card->dev, "cannot grab irq\n");
- snd_echo_free(chip);
return -EBUSY;
}
chip->irq = pci->irq;
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index be4d0489394a..0afe13f7b6e5 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -298,7 +298,12 @@ struct audiopipe {
* the current dma position
* (lower 32 bits only)
*/
- u32 last_counter; /* The last position, which is used
+ u32 last_period; /* Counter position last time a
+ * period elapsed
+ */
+ u32 last_counter; /* Used exclusively by pcm_pointer
+ * under PCM core locks.
+ * The last position, which is used
* to compute...
*/
u32 position; /* ...the number of bytes tranferred
@@ -332,11 +337,10 @@ struct audioformat {
struct echoaudio {
spinlock_t lock;
struct snd_pcm_substream *substream[DSP_MAXPIPES];
- int last_period[DSP_MAXPIPES];
struct mutex mode_mutex;
u16 num_digital_modes, digital_mode_list[6];
u16 num_clock_sources, clock_source_list[10];
- atomic_t opencount;
+ unsigned int opencount; /* protected by mode_mutex */
struct snd_kcontrol *clock_src_ctl;
struct snd_pcm *analog_pcm, *digital_pcm;
struct snd_card *card;
@@ -353,8 +357,8 @@ struct echoaudio {
struct timer_list timer;
char tinuse; /* Timer in use */
char midi_full; /* MIDI output buffer is full */
- char can_set_rate;
- char rate_set;
+ char can_set_rate; /* protected by mode_mutex */
+ char rate_set; /* protected by mode_mutex */
/* This stuff is used mainly by the lowlevel code */
struct comm_page *comm_page; /* Virtual address of the memory
@@ -415,7 +419,7 @@ struct echoaudio {
short asic_code; /* Current ASIC code */
u32 comm_page_phys; /* Physical address of the
* memory seen by DSP */
- volatile u32 __iomem *dsp_registers; /* DSP's register base */
+ u32 __iomem *dsp_registers; /* DSP's register base */
u32 active_mask; /* Chs. active mask or
* punks out */
#ifdef CONFIG_PM_SLEEP
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index f02f5b1568de..d10d0e460f0b 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -898,7 +898,7 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask)
return 0;
}
- dev_warn(chip->card->dev, "pause_transport: No pipes to stop!\n");
+ dev_dbg(chip->card->dev, "pause_transport: No pipes to stop!\n");
return 0;
}
@@ -924,7 +924,7 @@ static int stop_transport(struct echoaudio *chip, u32 channel_mask)
return 0;
}
- dev_warn(chip->card->dev, "stop_transport: No pipes to stop!\n");
+ dev_dbg(chip->card->dev, "stop_transport: No pipes to stop!\n");
return 0;
}
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
index dce9e57d01c4..f77db83dd73d 100644
--- a/sound/pci/echoaudio/mona_dsp.c
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -300,11 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock)
u32 control_reg, clocks_from_dsp;
int err;
-
- /* Prevent two simultaneous calls to switch_asic() */
- if (atomic_read(&chip->opencount))
- return -EAGAIN;
-
/* Mask off the clock select bits */
control_reg = le32_to_cpu(chip->comm_page->control_register) &
GML_CLOCK_CLEAR_MASK;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 6ff581733a19..bd70e112ffd7 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -623,7 +623,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 *emu)
static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu)
{
unsigned long special_port;
- unsigned int value;
+ __always_unused unsigned int value;
/* Special initialisation routine
* before the rest of the IO-Ports become active.
@@ -653,7 +653,7 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu,
int n, i;
int reg;
int value;
- unsigned int write_post;
+ __always_unused unsigned int write_post;
unsigned long flags;
if (!fw_entry)
diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c
index b3aa7bbe1067..89890f24509f 100644
--- a/sound/pci/emu10k1/emu10k1_patch.c
+++ b/sound/pci/emu10k1/emu10k1_patch.c
@@ -27,7 +27,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp,
const void __user *data, long count)
{
int offset;
- int truesize, size, loopsize, blocksize;
+ int truesize, size, blocksize;
+ __maybe_unused int loopsize;
int loopend, sampleend;
unsigned int start_addr;
struct snd_emu10k1 *emu;
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index b934c6ac52dd..b2ddabb99438 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_START:
snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */
snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]);
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
@@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream,
snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]);
}
snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra);
-
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index b4a0adf7451c..09704a78d799 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1619,7 +1619,8 @@ static int snd_es1938_create(struct snd_card *card,
static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
{
struct es1938 *chip = dev_id;
- unsigned char status, audiostatus;
+ unsigned char status;
+ __always_unused unsigned char audiostatus;
int handled = 0;
status = inb(SLIO_REG(chip, IRQCONTROL));
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index d26004b35a81..34332d008b27 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2631,7 +2631,7 @@ struct ess_device_list {
unsigned short vendor; /* subsystem vendor id */
};
-static const struct ess_device_list pm_whitelist[] = {
+static const struct ess_device_list pm_allowlist[] = {
{ TYPE_MAESTRO2E, 0x0e11 }, /* Compaq Armada */
{ TYPE_MAESTRO2E, 0x1028 },
{ TYPE_MAESTRO2E, 0x103c },
@@ -2642,7 +2642,7 @@ static const struct ess_device_list pm_whitelist[] = {
{ TYPE_MAESTRO2, 0x125d }, /* a PCI card, e.g. SF64-PCE2 */
};
-static const struct ess_device_list mpu_blacklist[] = {
+static const struct ess_device_list mpu_denylist[] = {
{ TYPE_MAESTRO2, 0x125d },
};
@@ -2724,12 +2724,12 @@ static int snd_es1968_create(struct snd_card *card,
pci_set_master(pci);
if (do_pm > 1) {
- /* disable power-management if not on the whitelist */
+ /* disable power-management if not on the allowlist */
unsigned short vend;
pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
- for (i = 0; i < (int)ARRAY_SIZE(pm_whitelist); i++) {
- if (chip->type == pm_whitelist[i].type &&
- vend == pm_whitelist[i].vendor) {
+ for (i = 0; i < (int)ARRAY_SIZE(pm_allowlist); i++) {
+ if (chip->type == pm_allowlist[i].type &&
+ vend == pm_allowlist[i].vendor) {
do_pm = 1;
break;
}
@@ -2848,12 +2848,12 @@ static int snd_es1968_probe(struct pci_dev *pci,
}
if (enable_mpu[dev] == 2) {
- /* check the black list */
+ /* check the deny list */
unsigned short vend;
pci_read_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, &vend);
- for (i = 0; i < ARRAY_SIZE(mpu_blacklist); i++) {
- if (chip->type == mpu_blacklist[i].type &&
- vend == mpu_blacklist[i].vendor) {
+ for (i = 0; i < ARRAY_SIZE(mpu_denylist); i++) {
+ if (chip->type == mpu_denylist[i].type &&
+ vend == mpu_denylist[i].vendor) {
enable_mpu[dev] = 0;
break;
}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 181ebafa550a..0a95032fd297 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -144,6 +144,8 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers");
/**
* struct fm801 - describes FM801 chip
+ * @dev: device for this chio
+ * @irq: irq number
* @port: I/O port number
* @multichannel: multichannel support
* @secondary: secondary codec
@@ -151,6 +153,31 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers");
* @tea575x_tuner: tuner access method & flags
* @ply_ctrl: playback control
* @cap_ctrl: capture control
+ * @ply_buffer: playback buffer
+ * @ply_buf: playback buffer index
+ * @ply_count: playback buffer count
+ * @ply_size: playback buffer size
+ * @ply_pos: playback position
+ * @cap_buffer: capture buffer
+ * @cap_buf: capture buffer index
+ * @cap_count: capture buffer count
+ * @cap_size: capture buffer size
+ * @cap_pos: capture position
+ * @ac97_bus: ac97 bus handle
+ * @ac97: ac97 handle
+ * @ac97_sec: ac97 secondary handle
+ * @card: ALSA card
+ * @pcm: PCM devices
+ * @rmidi: rmidi device
+ * @playback_substream: substream for playback
+ * @capture_substream: substream for capture
+ * @p_dma_size: playback DMA size
+ * @c_dma_size: capture DMA size
+ * @reg_lock: lock
+ * @proc_entry: /proc entry
+ * @v4l2_dev: v4l2 device
+ * @tea: tea575a structure
+ * @saved_regs: context saved during suspend
*/
struct fm801 {
struct device *dev;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 7ba542e45a3d..90759391cbac 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -8,6 +8,9 @@ config SND_HDA
select SND_JACK
select SND_HDA_CORE
+config SND_HDA_GENERIC_LEDS
+ bool
+
config SND_HDA_INTEL
tristate "HD Audio PCI"
depends on SND_PCI
@@ -91,6 +94,7 @@ config SND_HDA_PATCH_LOADER
config SND_HDA_CODEC_REALTEK
tristate "Build Realtek HD-audio codec support"
select SND_HDA_GENERIC
+ select SND_HDA_GENERIC_LEDS
help
Say Y or M here to include Realtek HD-audio codec support in
snd-hda-intel driver, such as ALC880.
@@ -111,6 +115,7 @@ comment "Set to Y if you want auto-loading the codec driver"
config SND_HDA_CODEC_SIGMATEL
tristate "Build IDT/Sigmatel HD-audio codec support"
select SND_HDA_GENERIC
+ select SND_HDA_GENERIC_LEDS
help
Say Y or M here to include IDT (Sigmatel) HD-audio codec support in
snd-hda-intel driver, such as STAC9200.
@@ -155,6 +160,7 @@ comment "Set to Y if you want auto-loading the codec driver"
config SND_HDA_CODEC_CONEXANT
tristate "Build Conexant HD-audio codec support"
select SND_HDA_GENERIC
+ select SND_HDA_GENERIC_LEDS
help
Say Y or M here to include Conexant HD-audio codec support in
snd-hda-intel driver, such as CX20549.
@@ -215,6 +221,10 @@ comment "Set to Y if you want auto-loading the codec driver"
config SND_HDA_GENERIC
tristate "Enable generic HD-audio codec parser"
+ select NEW_LEDS if SND_HDA_GENERIC_LEDS
+ select LEDS_CLASS if SND_HDA_GENERIC_LEDS
+ select LEDS_TRIGGERS if SND_HDA_GENERIC_LEDS
+ select LEDS_TRIGGER_AUDIO if SND_HDA_GENERIC_LEDS
help
Say Y or M here to enable the generic HD-audio codec parser
in snd-hda-intel driver.
@@ -230,6 +240,20 @@ config SND_HDA_POWER_SAVE_DEFAULT
The default time-out value in seconds for HD-audio automatic
power-save mode. 0 means to disable the power-save mode.
+config SND_HDA_INTEL_HDMI_SILENT_STREAM
+ bool "Enable Silent Stream always for HDMI"
+ depends on SND_HDA_INTEL
+ help
+ Intel hardware has a feature called 'silent stream', that
+ keeps external HDMI receiver's analog circuitry powered on
+ avoiding 2-3 sec silence during playback start. This mechanism
+ relies on setting channel_id as 0xf, sending info packet and
+ preventing codec D3 entry (increasing platform static power
+ consumption when HDMI receiver is plugged-in). 2-3 sec silence
+ at the playback start is expected whenever there is format change.
+ (default is 2 channel format).
+ Say Y to enable Silent Stream feature.
+
endif
endmenu
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 2c6d2becfe1a..824f4ac1a8ce 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -72,6 +72,12 @@ static int compare_input_type(const void *ap, const void *bp)
if (a->type != b->type)
return (int)(a->type - b->type);
+ /* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+ if (a->is_headset_mic && b->is_headphone_mic)
+ return -1; /* don't swap */
+ else if (a->is_headphone_mic && b->is_headset_mic)
+ return 1; /* swap */
+
/* In case one has boost and the other one has not,
pick the one with boost first. */
return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 841523f6b88d..53a2b89f8983 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
case SND_BELL:
if (hz)
hz = 1000;
- /* fallthru */
+ fallthrough;
case SND_TONE:
if (beep->linear_tone)
beep->tone = beep_linear_tone(beep, hz);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3576e2d8452f..e96a87f1b611 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -785,7 +785,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec)
snd_array_free(&codec->spdif_out);
snd_array_free(&codec->verbs);
codec->preset = NULL;
- codec->slave_dig_outs = NULL;
+ codec->follower_dig_outs = NULL;
codec->spdif_status_reset = 0;
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
@@ -1807,11 +1807,11 @@ int snd_hda_codec_reset(struct hda_codec *codec)
return 0;
}
-typedef int (*map_slave_func_t)(struct hda_codec *, void *, struct snd_kcontrol *);
+typedef int (*map_follower_func_t)(struct hda_codec *, void *, struct snd_kcontrol *);
-/* apply the function to all matching slave ctls in the mixer list */
-static int map_slaves(struct hda_codec *codec, const char * const *slaves,
- const char *suffix, map_slave_func_t func, void *data)
+/* apply the function to all matching follower ctls in the mixer list */
+static int map_followers(struct hda_codec *codec, const char * const *followers,
+ const char *suffix, map_follower_func_t func, void *data)
{
struct hda_nid_item *items;
const char * const *s;
@@ -1822,7 +1822,7 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
struct snd_kcontrol *sctl = items[i].kctl;
if (!sctl || sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
continue;
- for (s = slaves; *s; s++) {
+ for (s = followers; *s; s++) {
char tmpname[sizeof(sctl->id.name)];
const char *name = *s;
if (suffix) {
@@ -1841,8 +1841,8 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
return 0;
}
-static int check_slave_present(struct hda_codec *codec,
- void *data, struct snd_kcontrol *sctl)
+static int check_follower_present(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *sctl)
{
return 1;
}
@@ -1861,17 +1861,17 @@ static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
return 0;
}
-struct slave_init_arg {
+struct follower_init_arg {
struct hda_codec *codec;
int step;
};
-/* initialize the slave volume with 0dB via snd_ctl_apply_vmaster_slaves() */
-static int init_slave_0dB(struct snd_kcontrol *slave,
- struct snd_kcontrol *kctl,
- void *_arg)
+/* initialize the follower volume with 0dB via snd_ctl_apply_vmaster_followers() */
+static int init_follower_0dB(struct snd_kcontrol *follower,
+ struct snd_kcontrol *kctl,
+ void *_arg)
{
- struct slave_init_arg *arg = _arg;
+ struct follower_init_arg *arg = _arg;
int _tlv[4];
const int *tlv = NULL;
int step;
@@ -1880,7 +1880,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
if (kctl->tlv.c != snd_hda_mixer_amp_tlv) {
codec_err(arg->codec,
- "Unexpected TLV callback for slave %s:%d\n",
+ "Unexpected TLV callback for follower %s:%d\n",
kctl->id.name, kctl->id.index);
return 0; /* ignore */
}
@@ -1898,7 +1898,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
return 0;
if (arg->step && arg->step != step) {
codec_err(arg->codec,
- "Mismatching dB step for vmaster slave (%d!=%d)\n",
+ "Mismatching dB step for vmaster follower (%d!=%d)\n",
arg->step, step);
return 0;
}
@@ -1906,49 +1906,49 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
arg->step = step;
val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step;
if (val > 0) {
- put_kctl_with_value(slave, val);
+ put_kctl_with_value(follower, val);
return val;
}
return 0;
}
-/* unmute the slave via snd_ctl_apply_vmaster_slaves() */
-static int init_slave_unmute(struct snd_kcontrol *slave,
- struct snd_kcontrol *kctl,
- void *_arg)
+/* unmute the follower via snd_ctl_apply_vmaster_followers() */
+static int init_follower_unmute(struct snd_kcontrol *follower,
+ struct snd_kcontrol *kctl,
+ void *_arg)
{
- return put_kctl_with_value(slave, 1);
+ return put_kctl_with_value(follower, 1);
}
-static int add_slave(struct hda_codec *codec,
- void *data, struct snd_kcontrol *slave)
+static int add_follower(struct hda_codec *codec,
+ void *data, struct snd_kcontrol *follower)
{
- return snd_ctl_add_slave(data, slave);
+ return snd_ctl_add_follower(data, follower);
}
/**
- * __snd_hda_add_vmaster - create a virtual master control and add slaves
+ * __snd_hda_add_vmaster - create a virtual master control and add followers
* @codec: HD-audio codec
* @name: vmaster control name
* @tlv: TLV data (optional)
- * @slaves: slave control names (optional)
- * @suffix: suffix string to each slave name (optional)
- * @init_slave_vol: initialize slaves to unmute/0dB
+ * @followers: follower control names (optional)
+ * @suffix: suffix string to each follower name (optional)
+ * @init_follower_vol: initialize followers to unmute/0dB
* @ctl_ret: store the vmaster kcontrol in return
*
* Create a virtual master control with the given name. The TLV data
* must be either NULL or a valid data.
*
- * @slaves is a NULL-terminated array of strings, each of which is a
- * slave control name. All controls with these names are assigned to
+ * @followers is a NULL-terminated array of strings, each of which is a
+ * follower control name. All controls with these names are assigned to
* the new virtual master control.
*
* This function returns zero if successful or a negative error code.
*/
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves,
- const char *suffix, bool init_slave_vol,
+ unsigned int *tlv, const char * const *followers,
+ const char *suffix, bool init_follower_vol,
struct snd_kcontrol **ctl_ret)
{
struct snd_kcontrol *kctl;
@@ -1957,9 +1957,9 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (ctl_ret)
*ctl_ret = NULL;
- err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
+ err = map_followers(codec, followers, suffix, check_follower_present, NULL);
if (err != 1) {
- codec_dbg(codec, "No slave found for %s\n", name);
+ codec_dbg(codec, "No follower found for %s\n", name);
return 0;
}
kctl = snd_ctl_make_virtual_master(name, tlv);
@@ -1969,20 +1969,20 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
- err = map_slaves(codec, slaves, suffix, add_slave, kctl);
+ err = map_followers(codec, followers, suffix, add_follower, kctl);
if (err < 0)
return err;
/* init with master mute & zero volume */
put_kctl_with_value(kctl, 0);
- if (init_slave_vol) {
- struct slave_init_arg arg = {
+ if (init_follower_vol) {
+ struct follower_init_arg arg = {
.codec = codec,
.step = 0,
};
- snd_ctl_apply_vmaster_slaves(kctl,
- tlv ? init_slave_0dB : init_slave_unmute,
- &arg);
+ snd_ctl_apply_vmaster_followers(kctl,
+ tlv ? init_follower_0dB : init_follower_unmute,
+ &arg);
}
if (ctl_ret)
@@ -2285,7 +2285,7 @@ static unsigned int convert_to_spdif_status(unsigned short val)
return sbits;
}
-/* set digital convert verbs both for the given NID and its slaves */
+/* set digital convert verbs both for the given NID and its followers */
static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
int mask, int val)
{
@@ -2293,7 +2293,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
snd_hdac_regmap_update(&codec->core, nid, AC_VERB_SET_DIGI_CONVERT_1,
mask, val);
- d = codec->slave_dig_outs;
+ d = codec->follower_dig_outs;
if (!d)
return;
for (; *d; d++)
@@ -2936,6 +2936,10 @@ static int hda_codec_runtime_suspend(struct device *dev)
struct hda_codec *codec = dev_to_hda_codec(dev);
unsigned int state;
+ /* Nothing to do if card registration fails and the component driver never probes */
+ if (!codec->card)
+ return 0;
+
cancel_delayed_work_sync(&codec->jackpoll_work);
state = hda_call_codec_suspend(codec);
if (codec->link_down_at_suspend ||
@@ -2950,6 +2954,10 @@ static int hda_codec_runtime_resume(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
+ /* Nothing to do if card registration fails and the component driver never probes */
+ if (!codec->card)
+ return 0;
+
codec_display_power(codec, true);
snd_hdac_codec_link_up(&codec->core);
hda_call_codec_resume(codec);
@@ -3420,7 +3428,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save);
* @nid: NID to check / update
*
* Check whether the given NID is in the amp list. If it's in the list,
- * check the current AMP status, and update the the power-status according
+ * check the current AMP status, and update the power-status according
* to the mute status.
*
* This function is supposed to be set or called from the check_power_status
@@ -3581,9 +3589,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
spdif->ctls & ~AC_DIG1_ENABLE & 0xff,
-1);
snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
- if (codec->slave_dig_outs) {
+ if (codec->follower_dig_outs) {
const hda_nid_t *d;
- for (d = codec->slave_dig_outs; *d; d++)
+ for (d = codec->follower_dig_outs; *d; d++)
snd_hda_codec_setup_stream(codec, *d, stream_tag, 0,
format);
}
@@ -3596,9 +3604,9 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid)
{
snd_hda_codec_cleanup_stream(codec, nid);
- if (codec->slave_dig_outs) {
+ if (codec->follower_dig_outs) {
const hda_nid_t *d;
- for (d = codec->slave_dig_outs; *d; d++)
+ for (d = codec->follower_dig_outs; *d; d++)
snd_hda_codec_cleanup_stream(codec, *d);
}
}
@@ -3680,7 +3688,7 @@ EXPORT_SYMBOL_GPL(snd_hda_multi_out_dig_close);
* @hinfo: PCM information to assign
*
* Open analog outputs and set up the hw-constraints.
- * If the digital outputs can be opened as slave, open the digital
+ * If the digital outputs can be opened as follower, open the digital
* outputs, too.
*/
int snd_hda_multi_out_analog_open(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 9765652a73d7..80016b7b6849 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1202,15 +1202,8 @@ int azx_bus_init(struct azx *chip, const char *model)
if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY)
bus->core.align_bdle_4k = true;
- /* AMD chipsets often cause the communication stalls upon certain
- * sequence like the pin-detection. It seems that forcing the synced
- * access works around the stall. Grrr...
- */
- if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) {
- dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n");
- bus->core.sync_write = 1;
- bus->allow_bus_reset = 1;
- }
+ /* enable sync_write flag for stable communication as default */
+ bus->core.sync_write = 1;
return 0;
}
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 82e26442724b..be63ead8161f 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -33,7 +33,7 @@
#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */
#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 */
+/* 19 unused */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */
/* 22 unused */
@@ -41,7 +41,7 @@
/* 24 unused */
#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */
#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */
-/* 27 unused */
+#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */
#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */
#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */
#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index f4e9d9445e18..bbb17481159e 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -813,7 +813,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path,
}
}
-/* sync power of each widget in the the given path */
+/* sync power of each widget in the given path */
static hda_nid_t path_power_update(struct hda_codec *codec,
struct nid_path *path,
bool allow_powerdown)
@@ -3887,6 +3887,66 @@ static int parse_mic_boost(struct hda_codec *codec)
return 0;
}
+#ifdef CONFIG_SND_HDA_GENERIC_LEDS
+/*
+ * vmaster mute LED hook helpers
+ */
+
+static int create_mute_led_cdev(struct hda_codec *codec,
+ int (*callback)(struct led_classdev *,
+ enum led_brightness),
+ bool micmute)
+{
+ struct led_classdev *cdev;
+
+ cdev = devm_kzalloc(&codec->core.dev, sizeof(*cdev), GFP_KERNEL);
+ if (!cdev)
+ return -ENOMEM;
+
+ cdev->name = micmute ? "hda::micmute" : "hda::mute";
+ cdev->max_brightness = 1;
+ cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute";
+ cdev->brightness_set_blocking = callback;
+ cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE);
+ cdev->flags = LED_CORE_SUSPENDRESUME;
+
+ return devm_led_classdev_register(&codec->core.dev, cdev);
+}
+
+static void vmaster_update_mute_led(void *private_data, int enabled)
+{
+ ledtrig_audio_set(LED_AUDIO_MUTE, enabled ? LED_OFF : LED_ON);
+}
+
+/**
+ * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED
+ * @codec: the HDA codec
+ * @callback: the callback for LED classdev brightness_set_blocking
+ */
+int snd_hda_gen_add_mute_led_cdev(struct hda_codec *codec,
+ int (*callback)(struct led_classdev *,
+ enum led_brightness))
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int err;
+
+ if (callback) {
+ err = create_mute_led_cdev(codec, callback, false);
+ if (err) {
+ codec_warn(codec, "failed to create a mute LED cdev\n");
+ return err;
+ }
+ }
+
+ if (spec->vmaster_mute.hook)
+ codec_err(codec, "vmaster hook already present before cdev!\n");
+
+ spec->vmaster_mute.hook = vmaster_update_mute_led;
+ spec->vmaster_mute_enum = 1;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_mute_led_cdev);
+
/*
* mic mute LED hook helpers
*/
@@ -3921,8 +3981,8 @@ static void call_micmute_led_update(struct hda_codec *codec)
if (val == spec->micmute_led.led_value)
return;
spec->micmute_led.led_value = val;
- if (spec->micmute_led.update)
- spec->micmute_led.update(codec);
+ ledtrig_audio_set(LED_AUDIO_MICMUTE,
+ spec->micmute_led.led_value ? LED_ON : LED_OFF);
}
static void update_micmute_led(struct hda_codec *codec,
@@ -3994,20 +4054,8 @@ static const struct snd_kcontrol_new micmute_led_mode_ctl = {
.put = micmute_led_mode_put,
};
-/**
- * snd_hda_gen_add_micmute_led - helper for setting up mic mute LED hook
- * @codec: the HDA codec
- * @hook: the callback for updating LED
- *
- * Called from the codec drivers for offering the mic mute LED controls.
- * When established, it sets up cap_sync_hook and triggers the callback at
- * each time when the capture mixer switch changes. The callback is supposed
- * to update the LED accordingly.
- *
- * Returns 0 if the hook is established or a negative error code.
- */
-int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
- void (*hook)(struct hda_codec *))
+/* Set up the capture sync hook for controlling the mic-mute LED */
+static int add_micmute_led_hook(struct hda_codec *codec)
{
struct hda_gen_spec *spec = codec->spec;
@@ -4015,48 +4063,44 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
spec->micmute_led.capture = 0;
spec->micmute_led.led_value = 0;
spec->micmute_led.old_hook = spec->cap_sync_hook;
- spec->micmute_led.update = hook;
spec->cap_sync_hook = update_micmute_led;
if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl))
return -ENOMEM;
return 0;
}
-EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led);
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
-static void call_ledtrig_micmute(struct hda_codec *codec)
-{
- struct hda_gen_spec *spec = codec->spec;
-
- ledtrig_audio_set(LED_AUDIO_MICMUTE,
- spec->micmute_led.led_value ? LED_ON : LED_OFF);
-}
-#endif
/**
- * snd_hda_gen_fixup_micmute_led - A fixup for mic-mute LED trigger
- *
- * Pass this function to the quirk entry if another driver supports the
- * audio mic-mute LED trigger. Then this will bind the mixer capture switch
- * change with the LED.
+ * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED
+ * @codec: the HDA codec
+ * @callback: the callback for LED classdev brightness_set_blocking
*
- * Note that this fixup has to be called after other fixup that sets
- * cap_sync_hook. Otherwise the chaining wouldn't work.
+ * Called from the codec drivers for offering the mic mute LED controls.
+ * This creates a LED classdev and sets up the cap_sync_hook that is called at
+ * each time when the capture mixer switch changes.
*
- * @codec: the HDA codec
- * @fix: fixup pointer
- * @action: only supports HDA_FIXUP_ACT_PROBE value
+ * When NULL is passed to @callback, no classdev is created but only the
+ * LED-trigger is set up.
*
+ * Returns 0 or a negative error.
*/
-void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec,
- const struct hda_fixup *fix, int action)
+int snd_hda_gen_add_micmute_led_cdev(struct hda_codec *codec,
+ int (*callback)(struct led_classdev *,
+ enum led_brightness))
{
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
- if (action == HDA_FIXUP_ACT_PROBE)
- snd_hda_gen_add_micmute_led(codec, call_ledtrig_micmute);
-#endif
+ int err;
+
+ if (callback) {
+ err = create_mute_led_cdev(codec, callback, true);
+ if (err) {
+ codec_warn(codec, "failed to create a mic-mute LED cdev\n");
+ return err;
+ }
+ }
+
+ return add_micmute_led_hook(codec);
}
-EXPORT_SYMBOL_GPL(snd_hda_gen_fixup_micmute_led);
+EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led_cdev);
+#endif /* CONFIG_SND_HDA_GENERIC_LEDS */
/*
* parse digital I/Os and set up NIDs in BIOS auto-parse mode
@@ -4068,7 +4112,7 @@ static void parse_digital(struct hda_codec *codec)
int i, nums;
hda_nid_t dig_nid, pin;
- /* support multiple SPDIFs; the secondary is set up as a slave */
+ /* support multiple SPDIFs; the secondary is set up as a follower */
nums = 0;
for (i = 0; i < spec->autocfg.dig_outs; i++) {
pin = spec->autocfg.dig_out_pins[i];
@@ -4087,10 +4131,10 @@ static void parse_digital(struct hda_codec *codec)
spec->multiout.dig_out_nid = dig_nid;
spec->dig_out_type = spec->autocfg.dig_out_type[0];
} else {
- spec->multiout.slave_dig_outs = spec->slave_dig_outs;
- if (nums >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
+ spec->multiout.follower_dig_outs = spec->follower_dig_outs;
+ if (nums >= ARRAY_SIZE(spec->follower_dig_outs) - 1)
break;
- spec->slave_dig_outs[nums - 1] = dig_nid;
+ spec->follower_dig_outs[nums - 1] = dig_nid;
}
nums++;
}
@@ -4545,7 +4589,7 @@ static void call_update_outputs(struct hda_codec *codec)
else
snd_hda_gen_update_outputs(codec);
- /* sync the whole vmaster slaves to reflect the new auto-mute status */
+ /* sync the whole vmaster followers to reflect the new auto-mute status */
if (spec->auto_mute_via_amp && !codec->bus->shutdown)
snd_ctl_sync_vmaster(spec->vmaster_mute.sw_kctl, false);
}
@@ -5189,8 +5233,8 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_parse_auto_config);
* Build control elements
*/
-/* slave controls for virtual master */
-static const char * const slave_pfxs[] = {
+/* follower controls for virtual master */
+static const char * const follower_pfxs[] = {
"Front", "Surround", "Center", "LFE", "Side",
"Headphone", "Speaker", "Mono", "Line Out",
"CLFE", "Bass Speaker", "PCM",
@@ -5242,7 +5286,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
if (!spec->no_analog && !spec->suppress_vmaster &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- spec->vmaster_tlv, slave_pfxs,
+ spec->vmaster_tlv, follower_pfxs,
"Playback Volume");
if (err < 0)
return err;
@@ -5250,7 +5294,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
if (!spec->no_analog && !spec->suppress_vmaster &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_pfxs,
+ NULL, follower_pfxs,
"Playback Switch",
true, &spec->vmaster_mute.sw_kctl);
if (err < 0)
@@ -5765,7 +5809,7 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec)
spec->stream_name_digital);
if (!info)
return -ENOMEM;
- codec->slave_dig_outs = spec->multiout.slave_dig_outs;
+ codec->follower_dig_outs = spec->multiout.follower_dig_outs;
spec->pcm_rec[1] = info;
if (spec->dig_out_type)
info->pcm_type = spec->dig_out_type;
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index fb9f1a90238b..a43f0bb77dae 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -8,6 +8,8 @@
#ifndef __SOUND_HDA_GENERIC_H
#define __SOUND_HDA_GENERIC_H
+#include <linux/leds.h>
+
/* table entry for multi-io paths */
struct hda_multi_io {
hda_nid_t pin; /* multi-io widget pin NID */
@@ -86,7 +88,6 @@ struct hda_micmute_hook {
unsigned int led_mode;
unsigned int capture;
unsigned int led_value;
- void (*update)(struct hda_codec *codec);
void (*old_hook)(struct hda_codec *codec,
struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
@@ -115,7 +116,7 @@ struct hda_gen_spec {
* dig_out_nid and hp_nid are optional
*/
hda_nid_t alt_dac_nid;
- hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */
+ hda_nid_t follower_dig_outs[3]; /* optional - for auto-parsing */
int dig_out_type;
/* capture */
@@ -353,9 +354,11 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec,
void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on);
int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin);
-int snd_hda_gen_add_micmute_led(struct hda_codec *codec,
- void (*hook)(struct hda_codec *));
-void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec,
- const struct hda_fixup *fix, int action);
+int snd_hda_gen_add_mute_led_cdev(struct hda_codec *codec,
+ int (*callback)(struct led_classdev *,
+ enum led_brightness));
+int snd_hda_gen_add_micmute_led_cdev(struct hda_codec *codec,
+ int (*callback)(struct led_classdev *,
+ enum led_brightness));
#endif /* __SOUND_HDA_GENERIC_H */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d20aedd103c6..e34a4d5d047c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -180,7 +180,7 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout "
static bool pm_blacklist = true;
module_param(pm_blacklist, bool, 0644);
-MODULE_PARM_DESC(pm_blacklist, "Enable power-management blacklist");
+MODULE_PARM_DESC(pm_blacklist, "Enable power-management denylist");
/* reset the HD-audio controller in power save mode.
* this may give more power-saving, but will take longer time to
@@ -283,13 +283,12 @@ enum {
/* quirks for old Intel chipsets */
#define AZX_DCAPS_INTEL_ICH \
- (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\
- AZX_DCAPS_SYNC_WRITE)
+ (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE)
/* quirks for Intel PCH */
#define AZX_DCAPS_INTEL_PCH_BASE \
(AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
- AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+ AZX_DCAPS_SNOOP_TYPE(SCH))
/* PCH up to IVB; no runtime PM; bind with i915 gfx */
#define AZX_DCAPS_INTEL_PCH_NOPM \
@@ -298,19 +297,20 @@ enum {
/* PCH for HSW/BDW; with runtime PM */
/* no i915 binding for this as HSW/BDW has another controller for HDMI */
#define AZX_DCAPS_INTEL_PCH \
- (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME)
+ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+ AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
/* HSW HDMI */
#define AZX_DCAPS_INTEL_HASWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+ AZX_DCAPS_SNOOP_TYPE(SCH))
/* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
#define AZX_DCAPS_INTEL_BROADWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
+ AZX_DCAPS_SNOOP_TYPE(SCH))
#define AZX_DCAPS_INTEL_BAYTRAIL \
(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT)
@@ -321,19 +321,18 @@ 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_SKYLAKE
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
- (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\
+ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\
AZX_DCAPS_SNOOP_TYPE(ATI))
/* quirks for ATI/AMD HDMI */
#define AZX_DCAPS_PRESET_ATI_HDMI \
- (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB|\
+ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB|\
AZX_DCAPS_NO_MSI64)
/* quirks for ATI HDMI with snoop off */
@@ -342,7 +341,7 @@ enum {
/* quirks for AMD SB */
#define AZX_DCAPS_PRESET_AMD_SB \
- (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\
+ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\
AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
/* quirks for Nvidia */
@@ -1028,7 +1027,14 @@ static int azx_suspend(struct device *dev)
chip = card->private_data;
bus = azx_bus(chip);
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
- pm_runtime_force_suspend(dev);
+ /* An ugly workaround: direct call of __azx_runtime_suspend() and
+ * __azx_runtime_resume() for old Intel platforms that suffer from
+ * spurious wakeups after S3 suspend
+ */
+ if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+ __azx_runtime_suspend(chip);
+ else
+ pm_runtime_force_suspend(dev);
if (bus->irq >= 0) {
free_irq(bus->irq, chip);
bus->irq = -1;
@@ -1057,7 +1063,10 @@ static int azx_resume(struct device *dev)
if (azx_acquire_irq(chip, 1) < 0)
return -EIO;
- pm_runtime_force_resume(dev);
+ if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP)
+ __azx_runtime_resume(chip, false);
+ else
+ pm_runtime_force_resume(dev);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
trace_azx_resume(chip);
@@ -1508,7 +1517,7 @@ static bool check_hdmi_disabled(struct pci_dev *pci)
#endif /* SUPPORT_VGA_SWITCHEROO */
/*
- * white/black-listing for position_fix
+ * allow/deny-listing for position_fix
*/
static const struct snd_pci_quirk position_fix_list[] = {
SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
@@ -1601,7 +1610,7 @@ static void assign_position_fix(struct azx *chip, int fix)
}
/*
- * black-lists for probe_mask
+ * deny-lists for probe_mask
*/
static const struct snd_pci_quirk probe_mask_list[] = {
/* Thinkpad often breaks the controller communication when accessing
@@ -1649,9 +1658,9 @@ static void check_probe_mask(struct azx *chip, int dev)
}
/*
- * white/black-list for enable_msi
+ * allow/deny-list for enable_msi
*/
-static const struct snd_pci_quirk msi_black_list[] = {
+static const struct snd_pci_quirk msi_deny_list[] = {
SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */
SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */
SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */
@@ -1674,7 +1683,7 @@ static void check_msi(struct azx *chip)
return;
}
chip->msi = 1; /* enable MSI as default */
- q = snd_pci_quirk_lookup(chip->pci, msi_black_list);
+ q = snd_pci_quirk_lookup(chip->pci, msi_deny_list);
if (q) {
dev_info(chip->card->dev,
"msi for device %04x:%04x set to %d\n",
@@ -2074,11 +2083,11 @@ static void pcm_mmap_prepare(struct snd_pcm_substream *substream,
#endif
}
-/* Blacklist for skipping the whole probe:
+/* Denylist for skipping the whole probe:
* some HD-audio PCI entries are exposed without any codecs, and such devices
* should be ignored from the beginning.
*/
-static const struct pci_device_id driver_blacklist[] = {
+static const struct pci_device_id driver_denylist[] = {
{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */
{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */
{ PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */
@@ -2101,8 +2110,8 @@ static int azx_probe(struct pci_dev *pci,
bool schedule_probe;
int err;
- if (pci_match_id(driver_blacklist, pci)) {
- dev_info(&pci->dev, "Skipping the blacklisted device\n");
+ if (pci_match_id(driver_denylist, pci)) {
+ dev_info(&pci->dev, "Skipping the device on the denylist\n");
return -ENODEV;
}
@@ -2192,7 +2201,7 @@ out_free:
* So we keep a list of devices where we disable powersaving as its known
* to causes problems on these devices.
*/
-static const struct snd_pci_quirk power_save_blacklist[] = {
+static const struct snd_pci_quirk power_save_denylist[] = {
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
@@ -2238,9 +2247,9 @@ static void set_default_power_save(struct azx *chip)
if (pm_blacklist) {
const struct snd_pci_quirk *q;
- q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist);
+ q = snd_pci_quirk_lookup(chip->pci, power_save_denylist);
if (q && val) {
- dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n",
+ dev_info(chip->card->dev, "device %04x:%04x is on the power_save denylist, forcing power_save to 0\n",
q->subvendor, q->subdevice);
val = 0;
}
@@ -2343,7 +2352,6 @@ static int azx_probe_continue(struct azx *chip)
if (azx_has_pm_runtime(chip)) {
pm_runtime_use_autosuspend(&pci->dev);
- pm_runtime_allow(&pci->dev);
pm_runtime_put_autosuspend(&pci->dev);
}
@@ -2470,6 +2478,9 @@ static const struct pci_device_id azx_ids[] = {
/* Icelake */
{ PCI_DEVICE(0x8086, 0x34c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Icelake-H */
+ { PCI_DEVICE(0x8086, 0x3dc8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Jasperlake */
{ PCI_DEVICE(0x8086, 0x38c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
@@ -2478,9 +2489,14 @@ static const struct pci_device_id azx_ids[] = {
/* Tigerlake */
{ PCI_DEVICE(0x8086, 0xa0c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Tigerlake-H */
+ { PCI_DEVICE(0x8086, 0x43c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Elkhart Lake */
{ PCI_DEVICE(0x8086, 0x4b55),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ { PCI_DEVICE(0x8086, 0x4b58),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Broxton-P(Apollolake) */
{ PCI_DEVICE(0x8086, 0x5a98),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON },
@@ -2729,6 +2745,8 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI },
/* Zhaoxin */
{ PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN },
+ /* Loongson */
+ { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC },
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 3dca65d79b02..8c28b1022f49 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -129,11 +129,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name);
int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves,
- const char *suffix, bool init_slave_vol,
+ unsigned int *tlv, const char * const *followers,
+ const char *suffix, bool init_follower_vol,
struct snd_kcontrol **ctl_ret);
-#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \
- __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL)
+#define snd_hda_add_vmaster(codec, name, tlv, followers, suffix) \
+ __snd_hda_add_vmaster(codec, name, tlv, followers, suffix, true, NULL)
int snd_hda_codec_reset(struct hda_codec *codec);
void snd_hda_codec_register(struct hda_codec *codec);
void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
@@ -216,7 +216,7 @@ struct hda_multi_out {
hda_nid_t hp_out_nid[HDA_MAX_OUTS]; /* DACs for multiple HPs */
hda_nid_t extra_out_nid[HDA_MAX_OUTS]; /* other (e.g. speaker) DACs */
hda_nid_t dig_out_nid; /* digital out audio widget */
- const hda_nid_t *slave_dig_outs;
+ const hda_nid_t *follower_dig_outs;
int max_channels; /* currently supported analog channels */
int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */
int no_share_stream; /* don't share a stream with multiple pins */
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 0cc5fad1af8a..c94553bcca88 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -308,6 +308,7 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
return err;
}
bus->irq = irq_id;
+ bus->dma_stop_delay = 100;
card->sync_irq = bus->irq;
/*
@@ -333,6 +334,8 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
gcap = azx_readw(chip, GCAP);
dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap);
+ chip->align_buffer_size = 1;
+
/* read number of streams from GCAP register instead of using
* hardcoded value
*/
@@ -443,6 +446,7 @@ static int hda_tegra_create(struct snd_card *card,
if (err < 0)
return err;
+ chip->bus.core.sync_write = 0;
chip->bus.core.needs_damn_long_delay = 1;
chip->bus.core.aligned_mmio = 1;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 34fe753a46fb..b7dbf2e7f77a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1182,6 +1182,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+ SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
{}
};
@@ -4671,7 +4672,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
tmp = FLOAT_THREE;
break;
default:
@@ -4717,7 +4718,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
break;
default:
break;
@@ -4756,7 +4757,7 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
tmp = FLOAT_THREE;
break;
default:
@@ -5748,6 +5749,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
return 0;
}
+ if (nid == ZXR_HEADPHONE_GAIN) {
+ *valp = spec->zxr_gain_set;
+ return 0;
+ }
+
return 0;
}
@@ -6245,10 +6251,10 @@ static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
}
/*
- * Need to create slave controls for the alternate codecs that have surround
+ * Need to create follower controls for the alternate codecs that have surround
* capabilities.
*/
-static const char * const ca0132_alt_slave_pfxs[] = {
+static const char * const ca0132_alt_follower_pfxs[] = {
"Front", "Surround", "Center", "LFE", NULL,
};
@@ -6376,15 +6382,15 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
- /* Setup vmaster with surround slaves for desktop ca0132 devices */
+ /* Setup vmaster with surround followers for desktop ca0132 devices */
if (ca0132_use_alt_functions(spec)) {
snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT,
spec->tlv);
snd_hda_add_vmaster(codec, "Master Playback Volume",
- spec->tlv, ca0132_alt_slave_pfxs,
+ spec->tlv, ca0132_alt_follower_pfxs,
"Playback Volume");
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, ca0132_alt_slave_pfxs,
+ NULL, ca0132_alt_follower_pfxs,
"Playback Switch",
true, &spec->vmaster_mute.sw_kctl);
if (err < 0)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 396b5503038a..be5000dd1585 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -137,14 +137,16 @@ static void cx_auto_vmaster_hook(void *private_data, int enabled)
}
/* turn on/off EAPD according to Master switch (inversely!) for mute LED */
-static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled)
+static int cx_auto_vmaster_mute_led(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- struct hda_codec *codec = private_data;
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct conexant_spec *spec = codec->spec;
snd_hda_codec_write(codec, spec->mute_led_eapd, 0,
AC_VERB_SET_EAPD_BTLENABLE,
- enabled ? 0x00 : 0x02);
+ brightness ? 0x02 : 0x00);
+ return 0;
}
static int cx_auto_init(struct hda_codec *codec)
@@ -566,7 +568,7 @@ static void cxt_fixup_mute_led_eapd(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mute_led_eapd = 0x1b;
spec->dynamic_eapd = 1;
- spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook_mute_led;
+ snd_hda_gen_add_mute_led_cdev(codec, cx_auto_vmaster_mute_led);
}
}
@@ -631,21 +633,25 @@ static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask,
}
/* turn on/off mute LED via GPIO per vmaster hook */
-static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
+static int cxt_gpio_mute_update(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- struct hda_codec *codec = private_data;
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct conexant_spec *spec = codec->spec;
- /* muted -> LED on */
- cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, !enabled);
+
+ cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, brightness);
+ return 0;
}
/* turn on/off mic-mute LED via GPIO per capture hook */
-static void cxt_gpio_micmute_update(struct hda_codec *codec)
+static int cxt_gpio_micmute_update(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct conexant_spec *spec = codec->spec;
- cxt_update_gpio_led(codec, spec->gpio_mic_led_mask,
- spec->gen.micmute_led.led_value);
+ cxt_update_gpio_led(codec, spec->gpio_mic_led_mask, brightness);
+ return 0;
}
@@ -660,12 +666,12 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
};
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
- spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;
+ snd_hda_gen_add_mute_led_cdev(codec, cxt_gpio_mute_update);
spec->gpio_led = 0;
spec->mute_led_polarity = 0;
spec->gpio_mute_led_mask = 0x01;
spec->gpio_mic_led_mask = 0x02;
- snd_hda_gen_add_micmute_led(codec, cxt_gpio_micmute_update);
+ snd_hda_gen_add_micmute_led_cdev(codec, cxt_gpio_micmute_update);
}
snd_hda_add_verbs(codec, gpio_init);
if (spec->gpio_led)
@@ -988,8 +994,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
cx_auto_parse_eapd(codec);
spec->gen.own_eapd_ctl = 1;
- if (spec->dynamic_eapd)
- spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook;
switch (codec->core.vendor_id) {
case 0x14f15045:
@@ -1014,7 +1018,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
break;
case 0x14f150f2:
codec->power_save_node = 1;
- /* Fall through */
+ fallthrough;
default:
codec->pin_amp_workaround = 1;
snd_hda_pick_fixup(codec, cxt5066_fixup_models,
@@ -1022,17 +1026,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
break;
}
- /* Show mute-led control only on HP laptops
- * This is a sort of white-list: on HP laptops, EAPD corresponds
- * only to the mute-LED without actualy amp function. Meanwhile,
- * others may use EAPD really as an amp switch, so it might be
- * not good to expose it blindly.
- */
- switch (codec->core.subsystem_id >> 16) {
- case 0x103c:
- spec->gen.vmaster_mute_enum = 1;
- break;
- }
+ if (!spec->gen.vmaster_mute.hook && spec->dynamic_eapd)
+ spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook;
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index fbd7cc6026d8..b8c8490e568b 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -42,6 +42,11 @@ static bool enable_acomp = true;
module_param(enable_acomp, bool, 0444);
MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)");
+static bool enable_silent_stream =
+IS_ENABLED(CONFIG_SND_HDA_INTEL_HDMI_SILENT_STREAM);
+module_param(enable_silent_stream, bool, 0644);
+MODULE_PARM_DESC(enable_silent_stream, "Enable Silent Stream for HDMI devices");
+
struct hdmi_spec_per_cvt {
hda_nid_t cvt_nid;
int assigned;
@@ -160,6 +165,7 @@ struct hdmi_spec {
bool use_acomp_notifier; /* use eld_notify callback for hotplug */
bool acomp_registered; /* audio component registered in this driver */
+ bool force_connect; /* force connectivity */
struct drm_audio_component_audio_ops drm_audio_ops;
int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */
@@ -167,6 +173,7 @@ struct hdmi_spec {
hda_nid_t vendor_nid;
const int *port_map;
int port_num;
+ bool send_silent_stream; /* Flag to enable silent stream feature */
};
#ifdef CONFIG_SND_HDA_COMPONENT
@@ -259,7 +266,7 @@ static int hinfo_to_pcm_index(struct hda_codec *codec,
if (get_pcm_rec(spec, pcm_idx)->stream == hinfo)
return pcm_idx;
- codec_warn(codec, "HDMI: hinfo %p not registered\n", hinfo);
+ codec_warn(codec, "HDMI: hinfo %p not tied to a PCM\n", hinfo);
return -EINVAL;
}
@@ -277,7 +284,8 @@ static int hinfo_to_pin_index(struct hda_codec *codec,
return pin_idx;
}
- codec_dbg(codec, "HDMI: hinfo %p not registered\n", hinfo);
+ codec_dbg(codec, "HDMI: hinfo %p (pcm %d) not registered\n", hinfo,
+ hinfo_to_pcm_index(codec, hinfo));
return -EINVAL;
}
@@ -1634,21 +1642,72 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin,
snd_hda_power_down_pm(codec);
}
+static void silent_stream_enable(struct hda_codec *codec,
+ struct hdmi_spec_per_pin *per_pin)
+{
+ unsigned int newval, oldval;
+
+ codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n",
+ per_pin->pin_nid);
+
+ mutex_lock(&per_pin->lock);
+
+ if (!per_pin->channels)
+ per_pin->channels = 2;
+
+ oldval = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
+ AC_VERB_GET_CONV, 0);
+ newval = (oldval & 0xF0) | 0xF;
+ snd_hda_codec_write(codec, per_pin->pin_nid, 0,
+ AC_VERB_SET_CHANNEL_STREAMID, newval);
+
+ hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm);
+
+ mutex_unlock(&per_pin->lock);
+}
+
/* update ELD and jack state via audio component */
static void sync_eld_via_acomp(struct hda_codec *codec,
struct hdmi_spec_per_pin *per_pin)
{
struct hdmi_spec *spec = codec->spec;
struct hdmi_eld *eld = &spec->temp_eld;
+ bool monitor_prev, monitor_next;
mutex_lock(&per_pin->lock);
eld->monitor_present = false;
+ monitor_prev = per_pin->sink_eld.monitor_present;
eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid,
per_pin->dev_id, &eld->monitor_present,
eld->eld_buffer, ELD_MAX_SIZE);
eld->eld_valid = (eld->eld_size > 0);
update_eld(codec, per_pin, eld, 0);
+ monitor_next = per_pin->sink_eld.monitor_present;
mutex_unlock(&per_pin->lock);
+
+ /*
+ * Power-up will call hdmi_present_sense, so the PM calls
+ * have to be done without mutex held.
+ */
+
+ if (spec->send_silent_stream) {
+ int pm_ret;
+
+ if (!monitor_prev && monitor_next) {
+ pm_ret = snd_hda_power_up_pm(codec);
+ if (pm_ret < 0)
+ codec_err(codec,
+ "Monitor plugged-in, Failed to power up codec ret=[%d]\n",
+ pm_ret);
+ silent_stream_enable(codec, per_pin);
+ } else if (monitor_prev && !monitor_next) {
+ pm_ret = snd_hda_power_down_pm(codec);
+ if (pm_ret < 0)
+ codec_err(codec,
+ "Monitor plugged-out, Failed to power down codec ret=[%d]\n",
+ pm_ret);
+ }
+ }
}
static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
@@ -1700,7 +1759,8 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid)
* all device entries on the same pin
*/
config = snd_hda_codec_get_pincfg(codec, pin_nid);
- if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
+ if (get_defcfg_connect(config) == AC_JACK_PORT_NONE &&
+ !spec->force_connect)
return 0;
/*
@@ -1802,35 +1862,58 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
return 0;
}
+static const struct snd_pci_quirk force_connect_list[] = {
+ SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
+ SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+ {}
+};
+
static int hdmi_parse_codec(struct hda_codec *codec)
{
- hda_nid_t nid;
+ struct hdmi_spec *spec = codec->spec;
+ hda_nid_t start_nid;
+ unsigned int caps;
int i, nodes;
+ const struct snd_pci_quirk *q;
- nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
- if (!nid || nodes < 0) {
+ nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+ if (!start_nid || nodes < 0) {
codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
return -EINVAL;
}
- for (i = 0; i < nodes; i++, nid++) {
- unsigned int caps;
- unsigned int type;
+ q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list);
+
+ if (q && q->value)
+ spec->force_connect = true;
+
+ /*
+ * hdmi_add_pin() assumes total amount of converters to
+ * be known, so first discover all converters
+ */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
caps = get_wcaps(codec, nid);
- type = get_wcaps_type(caps);
if (!(caps & AC_WCAP_DIGITAL))
continue;
- switch (type) {
- case AC_WID_AUD_OUT:
+ if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
hdmi_add_cvt(codec, nid);
- break;
- case AC_WID_PIN:
+ }
+
+ /* discover audio pins */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
+
+ caps = get_wcaps(codec, nid);
+
+ if (!(caps & AC_WCAP_DIGITAL))
+ continue;
+
+ if (get_wcaps_type(caps) == AC_WID_PIN)
hdmi_add_pin(codec, nid);
- break;
- }
}
return 0;
@@ -2429,6 +2512,7 @@ static void generic_acomp_notifier_set(struct drm_audio_component *acomp,
mutex_lock(&spec->bind_lock);
spec->use_acomp_notifier = use_acomp;
spec->codec->relaxed_resume = use_acomp;
+ spec->codec->bus->keep_power = 0;
/* reprogram each jack detection logic depending on the notifier */
for (i = 0; i < spec->num_pins; i++)
reprogram_jack_detect(spec->codec,
@@ -2523,7 +2607,6 @@ static void generic_acomp_init(struct hda_codec *codec,
if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
match_bound_vga, 0)) {
spec->acomp_registered = true;
- codec->bus->keep_power = 0;
}
}
@@ -2791,6 +2874,13 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid,
spec->ops.setup_stream = i915_hsw_setup_stream;
spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup;
+ /*
+ * Enable silent stream feature, if it is enabled via
+ * module param or Kconfig option
+ */
+ if (enable_silent_stream)
+ spec->send_silent_stream = true;
+
return parse_intel_hdmi(codec);
}
@@ -4145,6 +4235,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch),
HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch),
HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d73f8beadb6..a1fa983d2a94 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -67,6 +67,13 @@ struct alc_customize_define {
unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */
};
+struct alc_coef_led {
+ unsigned int idx;
+ unsigned int mask;
+ unsigned int on;
+ unsigned int off;
+};
+
struct alc_spec {
struct hda_gen_spec gen; /* must be at head */
@@ -80,7 +87,7 @@ struct alc_spec {
unsigned int gpio_data;
bool gpio_write_delay; /* add a delay before writing gpio_data */
- /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
+ /* mute LED for HP laptops, see vref_mute_led_set() */
int mute_led_polarity;
int micmute_led_polarity;
hda_nid_t mute_led_nid;
@@ -88,14 +95,8 @@ struct alc_spec {
unsigned int gpio_mute_led_mask;
unsigned int gpio_mic_led_mask;
- unsigned int mute_led_coef_idx;
- unsigned int mute_led_coefbit_mask;
- unsigned int mute_led_coefbit_on;
- unsigned int mute_led_coefbit_off;
- unsigned int mic_led_coef_idx;
- unsigned int mic_led_coefbit_mask;
- unsigned int mic_led_coefbit_on;
- unsigned int mic_led_coefbit_off;
+ struct alc_coef_led mute_led_coef;
+ struct alc_coef_led mic_led_coef;
hda_nid_t headset_mic_pin;
hda_nid_t headphone_mic_pin;
@@ -287,6 +288,13 @@ static void alc_fixup_gpio4(struct hda_codec *codec,
alc_fixup_gpio(codec, action, 0x04);
}
+static void alc_fixup_micmute_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PROBE)
+ snd_hda_gen_add_micmute_led_cdev(codec, NULL);
+}
+
/*
* Fix hardware PLL issue
* On some codecs, the analog PLL gating control must be off while
@@ -374,7 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
case 0x10ec0295:
case 0x10ec0299:
alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000);
- /* fallthrough */
+ fallthrough;
case 0x10ec0215:
case 0x10ec0233:
case 0x10ec0235:
@@ -1070,7 +1078,7 @@ static int set_beep_amp(struct alc_spec *spec, hda_nid_t nid,
return 0;
}
-static const struct snd_pci_quirk beep_white_list[] = {
+static const struct snd_pci_quirk beep_allow_list[] = {
SND_PCI_QUIRK(0x1043, 0x103c, "ASUS", 1),
SND_PCI_QUIRK(0x1043, 0x115d, "ASUS", 1),
SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
@@ -1080,7 +1088,7 @@ static const struct snd_pci_quirk beep_white_list[] = {
SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1),
SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
- /* blacklist -- no beep available */
+ /* denylist -- no beep available */
SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0),
SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0),
{}
@@ -1090,7 +1098,7 @@ static inline int has_cdefine_beep(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
const struct snd_pci_quirk *q;
- q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list);
+ q = snd_pci_quirk_lookup(codec->bus->pci, beep_allow_list);
if (q)
return q->value;
return spec->cdefine.enable_pcbeep;
@@ -2461,6 +2469,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
@@ -3981,25 +3990,34 @@ static void alc269_fixup_x101_headset_mic(struct hda_codec *codec,
}
}
+static void alc_update_vref_led(struct hda_codec *codec, hda_nid_t pin,
+ bool polarity, bool on)
+{
+ unsigned int pinval;
+
+ if (!pin)
+ return;
+ if (polarity)
+ on = !on;
+ pinval = snd_hda_codec_get_pin_target(codec, pin);
+ pinval &= ~AC_PINCTL_VREFEN;
+ pinval |= on ? AC_PINCTL_VREF_80 : AC_PINCTL_VREF_HIZ;
+ /* temporarily power up/down for setting VREF */
+ snd_hda_power_up_pm(codec);
+ snd_hda_set_pin_ctl_cache(codec, pin, pinval);
+ snd_hda_power_down_pm(codec);
+}
/* update mute-LED according to the speaker mute state via mic VREF pin */
-static void alc269_fixup_mic_mute_hook(void *private_data, int enabled)
+static int vref_mute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- struct hda_codec *codec = private_data;
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct alc_spec *spec = codec->spec;
- unsigned int pinval;
- if (spec->mute_led_polarity)
- enabled = !enabled;
- pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid);
- pinval &= ~AC_PINCTL_VREFEN;
- pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80;
- if (spec->mute_led_nid) {
- /* temporarily power up/down for setting VREF */
- snd_hda_power_up_pm(codec);
- snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval);
- snd_hda_power_down_pm(codec);
- }
+ alc_update_vref_led(codec, spec->mute_led_nid,
+ spec->mute_led_polarity, brightness);
+ return 0;
}
/* Make sure the led works even in runtime suspend */
@@ -4037,8 +4055,7 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec,
break;
spec->mute_led_polarity = pol;
spec->mute_led_nid = pin - 0x0a + 0x18;
- spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
- spec->gen.vmaster_mute_enum = 1;
+ snd_hda_gen_add_mute_led_cdev(codec, vref_mute_led_set);
codec->power_filter = led_power_filter;
codec_dbg(codec,
"Detected mute LED for %x:%d\n", spec->mute_led_nid,
@@ -4056,8 +4073,7 @@ static void alc269_fixup_hp_mute_led_micx(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mute_led_polarity = 0;
spec->mute_led_nid = pin;
- spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook;
- spec->gen.vmaster_mute_enum = 1;
+ snd_hda_gen_add_mute_led_cdev(codec, vref_mute_led_set);
codec->power_filter = led_power_filter;
}
}
@@ -4090,26 +4106,18 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
}
/* turn on/off mute LED via GPIO per vmaster hook */
-static void alc_fixup_gpio_mute_hook(void *private_data, int enabled)
+static int gpio_mute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- struct hda_codec *codec = private_data;
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct alc_spec *spec = codec->spec;
alc_update_gpio_led(codec, spec->gpio_mute_led_mask,
- spec->mute_led_polarity, enabled);
+ spec->mute_led_polarity, !brightness);
+ return 0;
}
/* turn on/off mic-mute LED via GPIO per capture hook */
-static void alc_gpio_micmute_update(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
- spec->micmute_led_polarity,
- spec->gen.micmute_led.led_value);
-}
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
static int micmute_led_set(struct led_classdev *led_cdev,
enum led_brightness brightness)
{
@@ -4117,18 +4125,10 @@ static int micmute_led_set(struct led_classdev *led_cdev,
struct alc_spec *spec = codec->spec;
alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
- spec->micmute_led_polarity, !!brightness);
+ spec->micmute_led_polarity, !brightness);
return 0;
}
-static struct led_classdev micmute_led_cdev = {
- .name = "hda::micmute",
- .max_brightness = 1,
- .brightness_set_blocking = micmute_led_set,
- .default_trigger = "audio-micmute",
-};
-#endif
-
/* setup mute and mic-mute GPIO bits, add hooks appropriately */
static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
int action,
@@ -4136,9 +4136,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
unsigned int micmute_mask)
{
struct alc_spec *spec = codec->spec;
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
- int err;
-#endif
alc_fixup_gpio(codec, action, mute_mask | micmute_mask);
@@ -4146,18 +4143,11 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec,
return;
if (mute_mask) {
spec->gpio_mute_led_mask = mute_mask;
- spec->gen.vmaster_mute.hook = alc_fixup_gpio_mute_hook;
+ snd_hda_gen_add_mute_led_cdev(codec, gpio_mute_led_set);
}
if (micmute_mask) {
spec->gpio_mic_led_mask = micmute_mask;
- snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update);
-
-#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
- micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE);
- err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev);
- if (err)
- codec_warn(codec, "failed to register micmute LED\n");
-#endif
+ snd_hda_gen_add_micmute_led_cdev(codec, micmute_led_set);
}
}
@@ -4170,10 +4160,6 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec,
static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
- struct alc_spec *spec = codec->spec;
-
- spec->micmute_led_polarity = 1;
-
alc_fixup_hp_gpio_led(codec, action, 0x04, 0x01);
}
@@ -4183,21 +4169,16 @@ static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
alc_fixup_hp_gpio_led(codec, action, 0x02, 0x20);
}
-/* turn on/off mic-mute LED per capture hook */
-static void alc_cap_micmute_update(struct hda_codec *codec)
+/* turn on/off mic-mute LED per capture hook via VREF change */
+static int vref_micmute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct alc_spec *spec = codec->spec;
- unsigned int pinval;
- if (!spec->cap_mute_led_nid)
- return;
- pinval = snd_hda_codec_get_pin_target(codec, spec->cap_mute_led_nid);
- pinval &= ~AC_PINCTL_VREFEN;
- if (spec->gen.micmute_led.led_value)
- pinval |= AC_PINCTL_VREF_80;
- else
- pinval |= AC_PINCTL_VREF_HIZ;
- snd_hda_set_pin_ctl_cache(codec, spec->cap_mute_led_nid, pinval);
+ alc_update_vref_led(codec, spec->cap_mute_led_nid,
+ spec->micmute_led_polarity, brightness);
+ return 0;
}
static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
@@ -4213,7 +4194,7 @@ static void alc269_fixup_hp_gpio_mic1_led(struct hda_codec *codec,
spec->gpio_mask |= 0x10;
spec->gpio_dir |= 0x10;
spec->cap_mute_led_nid = 0x18;
- snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+ snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
codec->power_filter = led_power_filter;
}
}
@@ -4226,25 +4207,32 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec,
alc_fixup_hp_gpio_led(codec, action, 0x08, 0);
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->cap_mute_led_nid = 0x18;
- snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+ snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
codec->power_filter = led_power_filter;
}
}
+static void alc_update_coef_led(struct hda_codec *codec,
+ struct alc_coef_led *led,
+ bool polarity, bool on)
+{
+ if (polarity)
+ on = !on;
+ /* temporarily power up/down for setting COEF bit */
+ alc_update_coef_idx(codec, led->idx, led->mask,
+ on ? led->on : led->off);
+}
+
/* update mute-LED according to the speaker mute state via COEF bit */
-static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled)
+static int coef_mute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- struct hda_codec *codec = private_data;
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct alc_spec *spec = codec->spec;
- if (spec->mute_led_polarity)
- enabled = !enabled;
-
- /* temporarily power up/down for setting COEF bit */
- enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx,
- spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) :
- alc_update_coef_idx(codec, spec->mute_led_coef_idx,
- spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on);
+ alc_update_coef_led(codec, &spec->mute_led_coef,
+ spec->mute_led_polarity, brightness);
+ return 0;
}
static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
@@ -4255,12 +4243,11 @@ static void alc285_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mute_led_polarity = 0;
- spec->mute_led_coef_idx = 0x0b;
- spec->mute_led_coefbit_mask = 1<<3;
- spec->mute_led_coefbit_on = 1<<3;
- spec->mute_led_coefbit_off = 0;
- spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
- spec->gen.vmaster_mute_enum = 1;
+ spec->mute_led_coef.idx = 0x0b;
+ spec->mute_led_coef.mask = 1 << 3;
+ spec->mute_led_coef.on = 1 << 3;
+ spec->mute_led_coef.off = 0;
+ snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set);
}
}
@@ -4272,26 +4259,24 @@ static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec,
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->mute_led_polarity = 0;
- spec->mute_led_coef_idx = 0x34;
- spec->mute_led_coefbit_mask = 1<<5;
- spec->mute_led_coefbit_on = 0;
- spec->mute_led_coefbit_off = 1<<5;
- spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
- spec->gen.vmaster_mute_enum = 1;
+ spec->mute_led_coef.idx = 0x34;
+ spec->mute_led_coef.mask = 1 << 5;
+ spec->mute_led_coef.on = 0;
+ spec->mute_led_coef.off = 1 << 5;
+ snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set);
}
}
/* turn on/off mic-mute LED per capture hook by coef bit */
-static void alc_hp_cap_micmute_update(struct hda_codec *codec)
+static int coef_micmute_led_set(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct alc_spec *spec = codec->spec;
- if (spec->gen.micmute_led.led_value)
- alc_update_coef_idx(codec, spec->mic_led_coef_idx,
- spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on);
- else
- alc_update_coef_idx(codec, spec->mic_led_coef_idx,
- spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off);
+ alc_update_coef_led(codec, &spec->mic_led_coef,
+ spec->micmute_led_polarity, brightness);
+ return 0;
}
static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
@@ -4300,11 +4285,11 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
- spec->mic_led_coef_idx = 0x19;
- spec->mic_led_coefbit_mask = 1<<13;
- spec->mic_led_coefbit_on = 1<<13;
- spec->mic_led_coefbit_off = 0;
- snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+ spec->mic_led_coef.idx = 0x19;
+ spec->mic_led_coef.mask = 1 << 13;
+ spec->mic_led_coef.on = 1 << 13;
+ spec->mic_led_coef.off = 0;
+ snd_hda_gen_add_micmute_led_cdev(codec, coef_micmute_led_set);
}
}
@@ -4314,11 +4299,11 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
- spec->mic_led_coef_idx = 0x35;
- spec->mic_led_coefbit_mask = 3<<2;
- spec->mic_led_coefbit_on = 2<<2;
- spec->mic_led_coefbit_off = 1<<2;
- snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+ spec->mic_led_coef.idx = 0x35;
+ spec->mic_led_coef.mask = 3 << 2;
+ spec->mic_led_coef.on = 2 << 2;
+ spec->mic_led_coef.off = 1 << 2;
+ snd_hda_gen_add_micmute_led_cdev(codec, coef_micmute_led_set);
}
}
@@ -4458,7 +4443,7 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec,
alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1a);
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->cap_mute_led_nid = 0x18;
- snd_hda_gen_add_micmute_led(codec, alc_cap_micmute_update);
+ snd_hda_gen_add_micmute_led_cdev(codec, vref_micmute_led_set);
}
}
@@ -4709,7 +4694,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
break;
case 0x10ec0867:
alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14);
- /* fallthru */
+ fallthrough;
case 0x10ec0221:
case 0x10ec0662:
snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
@@ -5974,6 +5959,16 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
}
+static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action != HDA_FIXUP_ACT_INIT)
+ return;
+
+ msleep(100);
+ alc_write_coef_idx(codec, 0x65, 0x0);
+}
+
/* for hda_fixup_thinkpad_acpi() */
#include "thinkpad_helper.c"
@@ -6148,6 +6143,19 @@ enum {
ALC236_FIXUP_HP_MUTE_LED,
ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+ ALC269VC_FIXUP_ACER_HEADSET_MIC,
+ ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
+ ALC289_FIXUP_ASUS_GA401,
+ ALC289_FIXUP_ASUS_GA502,
+ ALC256_FIXUP_ACER_MIC_NO_PRESENCE,
+ ALC285_FIXUP_HP_GPIO_AMP_INIT,
+ ALC269_FIXUP_CZC_B20,
+ ALC269_FIXUP_CZC_TMI,
+ ALC269_FIXUP_CZC_L101,
+ ALC269_FIXUP_LEMOTE_A1802,
+ ALC269_FIXUP_LEMOTE_A190X,
+ ALC256_FIXUP_INTEL_NUC8_RUGGED,
};
static const struct hda_fixup alc269_fixups[] = {
@@ -6688,7 +6696,7 @@ static const struct hda_fixup alc269_fixups[] = {
},
[ALC255_FIXUP_MIC_MUTE_LED] = {
.type = HDA_FIXUP_FUNC,
- .v.func = snd_hda_gen_fixup_micmute_led,
+ .v.func = alc_fixup_micmute_led,
},
[ALC282_FIXUP_ASPIRE_V5_PINS] = {
.type = HDA_FIXUP_PINS,
@@ -6791,7 +6799,7 @@ static const struct hda_fixup alc269_fixups[] = {
},
[ALC292_FIXUP_DELL_E7X] = {
.type = HDA_FIXUP_FUNC,
- .v.func = snd_hda_gen_fixup_micmute_led,
+ .v.func = alc_fixup_micmute_led,
/* micmute fixup must be applied at last */
.chained_before = true,
.chain_id = ALC292_FIXUP_DELL_E7X_AAMIX,
@@ -7113,7 +7121,7 @@ static const struct hda_fixup alc269_fixups[] = {
{ }
},
.chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
},
[ALC294_FIXUP_ASUS_HEADSET_MIC] = {
.type = HDA_FIXUP_PINS,
@@ -7122,7 +7130,7 @@ static const struct hda_fixup alc269_fixups[] = {
{ }
},
.chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
},
[ALC294_FIXUP_ASUS_SPK] = {
.type = HDA_FIXUP_VERBS,
@@ -7130,6 +7138,8 @@ static const struct hda_fixup alc269_fixups[] = {
/* Set EAPD high */
{ 0x20, AC_VERB_SET_COEF_INDEX, 0x40 },
{ 0x20, AC_VERB_SET_PROC_COEF, 0x8800 },
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
{ }
},
.chained = true,
@@ -7326,6 +7336,156 @@ static const struct hda_fixup alc269_fixups[] = {
.chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE
},
+ [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x14, 0x90100120 }, /* use as internal speaker */
+ { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x01011020 }, /* use as line out */
+ { },
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x02a11030 }, /* use as headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC289_FIXUP_ASUS_GA401] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a11020 }, /* headset mic with jack detect */
+ { }
+ },
+ },
+ [ALC289_FIXUP_ASUS_GA502] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a11020 }, /* headset mic with jack detect */
+ { }
+ },
+ },
+ [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+ },
+ [ALC285_FIXUP_HP_GPIO_AMP_INIT] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_gpio_amp_init,
+ .chained = true,
+ .chain_id = ALC285_FIXUP_HP_GPIO_LED
+ },
+ [ALC269_FIXUP_CZC_B20] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x411111f0 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x15, 0x032f1020 }, /* HP out */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x03ab1040 }, /* mic */
+ { 0x19, 0xb7a7013f },
+ { 0x1a, 0x0181305f },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x411111f0 },
+ { 0x1e, 0x411111f0 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_CZC_TMI] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x4000c000 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x15, 0x0421401f }, /* HP out */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x04a19020 }, /* mic */
+ { 0x19, 0x411111f0 },
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x40448505 },
+ { 0x1e, 0x411111f0 },
+ { 0x20, 0x8000ffff },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_CZC_L101] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x40000000 },
+ { 0x14, 0x01014010 }, /* speaker */
+ { 0x15, 0x411111f0 }, /* HP out */
+ { 0x16, 0x411111f0 },
+ { 0x18, 0x01a19020 }, /* mic */
+ { 0x19, 0x02a19021 },
+ { 0x1a, 0x0181302f },
+ { 0x1b, 0x0221401f },
+ { 0x1c, 0x411111f0 },
+ { 0x1d, 0x4044c601 },
+ { 0x1e, 0x411111f0 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_LEMOTE_A1802] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x40000000 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x03a19040 }, /* mic1 */
+ { 0x19, 0x90a70130 }, /* mic2 */
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x40489d2d },
+ { 0x1e, 0x411111f0 },
+ { 0x20, 0x0003ffff },
+ { 0x21, 0x03214020 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_LEMOTE_A190X] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x14, 0x99130110 }, /* speaker */
+ { 0x15, 0x0121401f }, /* HP out */
+ { 0x18, 0x01a19c20 }, /* rear mic */
+ { 0x19, 0x99a3092f }, /* front mic */
+ { 0x1b, 0x0201401f }, /* front lineout */
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC256_FIXUP_INTEL_NUC8_RUGGED] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7341,16 +7501,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+ SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+ SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -7470,7 +7634,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
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(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -7492,6 +7658,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE),
SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
@@ -7501,6 +7668,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+ SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
+ SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -7520,11 +7689,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC),
SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE),
+ SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE),
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
+ SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
@@ -7568,8 +7741,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
- SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Yoga 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
- SND_PCI_QUIRK(0x17aa, 0x2293, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
@@ -7605,9 +7777,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
+ SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20),
+ SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
+ SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
+ SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
+ SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
#if 0
/* Below is a quirk table taken from the old code.
@@ -7779,6 +7957,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
+ {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
{}
};
#define ALC225_STANDARD_PINS \
@@ -8863,6 +9042,7 @@ enum {
ALC662_FIXUP_LED_GPIO1,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
+ ALC662_FIXUP_CZC_ET26,
ALC662_FIXUP_CZC_P10T,
ALC662_FIXUP_SKU_IGNORE,
ALC662_FIXUP_HP_RP5800,
@@ -8932,6 +9112,25 @@ static const struct hda_fixup alc662_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc272_fixup_mario,
},
+ [ALC662_FIXUP_CZC_ET26] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ {0x12, 0x403cc000},
+ {0x14, 0x90170110}, /* speaker */
+ {0x15, 0x411111f0},
+ {0x16, 0x411111f0},
+ {0x18, 0x01a19030}, /* mic */
+ {0x19, 0x90a7013f}, /* int-mic */
+ {0x1a, 0x01014020},
+ {0x1b, 0x0121401f},
+ {0x1c, 0x411111f0},
+ {0x1d, 0x411111f0},
+ {0x1e, 0x40478e35},
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_SKU_IGNORE
+ },
[ALC662_FIXUP_CZC_P10T] = {
.type = HDA_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -9315,6 +9514,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68),
SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON),
+ SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26),
SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a608d0486ae4..c662431bf13a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -320,15 +320,18 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask,
}
/* hook for controlling mic-mute LED GPIO */
-static void stac_capture_led_update(struct hda_codec *codec)
+static int stac_capture_led_update(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
struct sigmatel_spec *spec = codec->spec;
- if (spec->gen.micmute_led.led_value)
+ if (brightness)
spec->gpio_data |= spec->mic_mute_led_gpio;
else
spec->gpio_data &= ~spec->mic_mute_led_gpio;
stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
+ return 0;
}
static int stac_vrefout_set(struct hda_codec *codec,
@@ -366,10 +369,9 @@ static unsigned int stac_vref_led_power_filter(struct hda_codec *codec,
}
/* update mute-LED accoring to the master switch */
-static void stac_update_led_status(struct hda_codec *codec, int enabled)
+static void stac_update_led_status(struct hda_codec *codec, bool muted)
{
struct sigmatel_spec *spec = codec->spec;
- int muted = !enabled;
if (!spec->gpio_led)
return;
@@ -393,9 +395,13 @@ static void stac_update_led_status(struct hda_codec *codec, int enabled)
}
/* vmaster hook to update mute LED */
-static void stac_vmaster_hook(void *private_data, int val)
+static int stac_vmaster_hook(struct led_classdev *led_cdev,
+ enum led_brightness brightness)
{
- stac_update_led_status(private_data, val);
+ struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent);
+
+ stac_update_led_status(codec, brightness);
+ return 0;
}
/* automute hook to handle GPIO mute and EAPD updates */
@@ -832,7 +838,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec,
static const struct snd_kcontrol_new beep_vol_ctl =
HDA_CODEC_VOLUME(NULL, 0, 0, 0);
- /* check for mute support for the the amp */
+ /* check for mute support for the amp */
if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
const struct snd_kcontrol_new *temp;
if (spec->anabeep_nid == nid)
@@ -3129,7 +3135,7 @@ static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin)
unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin);
/* It was changed in the BIOS to just satisfy MS DTM.
- * Lets turn it back into slaved HP
+ * Lets turn it back into follower HP
*/
pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) |
(AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT);
@@ -4313,7 +4319,7 @@ static int stac_parse_auto_config(struct hda_codec *codec)
#endif
if (spec->gpio_led)
- spec->gen.vmaster_mute.hook = stac_vmaster_hook;
+ snd_hda_gen_add_mute_led_cdev(codec, stac_vmaster_hook);
if (spec->aloopback_ctl &&
snd_hda_get_bool_hint(codec, "loopback") == 1) {
@@ -4636,7 +4642,7 @@ static void stac_setup_gpio(struct hda_codec *codec)
spec->gpio_dir |= spec->mic_mute_led_gpio;
spec->mic_enabled = 0;
spec->gpio_data |= spec->mic_mute_led_gpio;
- snd_hda_gen_add_micmute_led(codec, stac_capture_led_update);
+ snd_hda_gen_add_micmute_led_cdev(codec, stac_capture_led_update);
}
}
diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c
index 4089feb8c68e..6698ae241efc 100644
--- a/sound/pci/hda/thinkpad_helper.c
+++ b/sound/pci/hda/thinkpad_helper.c
@@ -3,13 +3,11 @@
* to be included from codec driver
*/
-#if IS_ENABLED(CONFIG_THINKPAD_ACPI) && IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO)
+#if IS_ENABLED(CONFIG_THINKPAD_ACPI)
#include <linux/acpi.h>
#include <linux/leds.h>
-static void (*old_vmaster_hook)(void *, int);
-
static bool is_thinkpad(struct hda_codec *codec)
{
return (codec->core.subsystem_id >> 16 == 0x17aa) &&
@@ -17,25 +15,14 @@ static bool is_thinkpad(struct hda_codec *codec)
acpi_dev_found("IBM0068"));
}
-static void update_tpacpi_mute_led(void *private_data, int enabled)
-{
- if (old_vmaster_hook)
- old_vmaster_hook(private_data, enabled);
-
- ledtrig_audio_set(LED_AUDIO_MUTE, enabled ? LED_OFF : LED_ON);
-}
-
static void hda_fixup_thinkpad_acpi(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
- struct hda_gen_spec *spec = codec->spec;
-
if (action == HDA_FIXUP_ACT_PROBE) {
if (!is_thinkpad(codec))
return;
- old_vmaster_hook = spec->vmaster_mute.hook;
- spec->vmaster_mute.hook = update_tpacpi_mute_led;
- snd_hda_gen_fixup_micmute_led(codec, fix, action);
+ snd_hda_gen_add_mute_led_cdev(codec, NULL);
+ snd_hda_gen_add_micmute_led_cdev(codec, NULL);
}
}
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 81929063b2fa..1d2a0287284b 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice)
break;
case ICE1712_SUBDEVICE_DELTADIO2496:
ice->gpio.set_pro_rate = delta_1010_set_rate_val;
- /* fall thru */
+ fallthrough;
case ICE1712_SUBDEVICE_DELTA66:
ice->spdif.ops.open = delta_open_spdif;
ice->spdif.ops.setup_rate = delta_setup_spdif;
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 7be4eb42f05e..e57a55cebc5a 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -397,7 +397,7 @@ static const struct snd_kcontrol_new juli_mute_controls[] = {
},
};
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
PCM_VOLUME,
MONITOR_AN_IN_VOLUME,
MONITOR_DIG_IN_VOLUME,
@@ -418,16 +418,16 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card,
return snd_ctl_find_id(card, &sid);
}
-static void add_slaves(struct snd_card *card,
- struct snd_kcontrol *master,
- const char * const *list)
+static void add_followers(struct snd_card *card,
+ struct snd_kcontrol *master,
+ const char * const *list)
{
for (; *list; list++) {
- struct snd_kcontrol *slave = ctl_find(card, *list);
- /* dev_dbg(card->dev, "add_slaves - %s\n", *list); */
- if (slave) {
- /* dev_dbg(card->dev, "slave %s found\n", *list); */
- snd_ctl_add_slave(master, slave);
+ struct snd_kcontrol *follower = ctl_find(card, *list);
+ /* dev_dbg(card->dev, "add_followers - %s\n", *list); */
+ if (follower) {
+ /* dev_dbg(card->dev, "follower %s found\n", *list); */
+ snd_ctl_add_follower(master, follower);
}
}
}
@@ -454,7 +454,7 @@ static int juli_add_controls(struct snd_ice1712 *ice)
juli_master_db_scale);
if (!vmaster)
return -ENOMEM;
- add_slaves(ice->card, vmaster, slave_vols);
+ add_followers(ice->card, vmaster, follower_vols);
err = snd_ctl_add(ice->card, vmaster);
if (err < 0)
return err;
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 8df14f63b10d..096ec76f5304 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -32,7 +32,7 @@
* Experimentally I found out that only a combination of
* OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
* VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
- * sampling rate. That means the the FPGA doubles the
+ * sampling rate. That means that the FPGA doubles the
* MCK01 rate.
*
* Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c
index 866596205710..0e3e04aa9faf 100644
--- a/sound/pci/ice1712/quartet.c
+++ b/sound/pci/ice1712/quartet.c
@@ -757,7 +757,7 @@ static const struct snd_kcontrol_new qtet_controls[] = {
QTET_CONTROL("Output 3/4 to Monitor 1/2", sw, OUT34_MON12),
};
-static const char * const slave_vols[] = {
+static const char * const follower_vols[] = {
PCM_12_PLAYBACK_VOLUME,
PCM_34_PLAYBACK_VOLUME,
NULL
@@ -776,13 +776,13 @@ static struct snd_kcontrol *ctl_find(struct snd_card *card,
return snd_ctl_find_id(card, &sid);
}
-static void add_slaves(struct snd_card *card,
- struct snd_kcontrol *master, const char * const *list)
+static void add_followers(struct snd_card *card,
+ struct snd_kcontrol *master, const char * const *list)
{
for (; *list; list++) {
- struct snd_kcontrol *slave = ctl_find(card, *list);
- if (slave)
- snd_ctl_add_slave(master, slave);
+ struct snd_kcontrol *follower = ctl_find(card, *list);
+ if (follower)
+ snd_ctl_add_follower(master, follower);
}
}
@@ -806,7 +806,7 @@ static int qtet_add_controls(struct snd_ice1712 *ice)
qtet_master_db_scale);
if (!vmaster)
return -ENOMEM;
- add_slaves(ice->card, vmaster, slave_vols);
+ add_followers(ice->card, vmaster, follower_vols);
err = snd_ctl_add(ice->card, vmaster);
if (err < 0)
return err;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 1781a1c081c3..3349e455a871 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -66,7 +66,7 @@ MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard.");
module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for Intel i8x0 soundcard.");
module_param(ac97_clock, int, 0444);
-MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = whitelist + auto-detect, 1 = force autodetect).");
+MODULE_PARM_DESC(ac97_clock, "AC'97 codec clock (0 = allowlist + auto-detect, 1 = force autodetect).");
module_param(ac97_quirk, charp, 0444);
MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
module_param(buggy_semaphore, bool, 0444);
@@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
ichdev->suspended = 0;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
val = ICH_IOCE | ICH_STARTBM;
@@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
ichdev->suspended = 1;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_STOP:
val = 0;
break;
@@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
ichdev->suspended = 0;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
ichdev->suspended = 1;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
/* pause */
@@ -2792,7 +2792,7 @@ static int intel8x0_in_clock_list(struct intel8x0 *chip)
wl = snd_pci_quirk_lookup(pci, intel8x0_clock_list);
if (!wl)
return 0;
- dev_info(chip->card->dev, "white list rate for %04x:%04x is %i\n",
+ dev_info(chip->card->dev, "allow list rate for %04x:%04x is %i\n",
pci->subsystem_vendor, pci->subsystem_device, wl->value);
chip->ac97_bus->clock = wl->value;
return 1;
@@ -3138,7 +3138,7 @@ static const struct snd_pci_quirk spdif_aclink_defaults[] = {
{ } /* end */
};
-/* look up white/black list for SPDIF over ac-link */
+/* look up allow/deny list for SPDIF over ac-link */
static int check_default_spdif_aclink(struct pci_dev *pci)
{
const struct snd_pci_quirk *w;
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 65a887b217ee..2eddd9de9e6d 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2149,7 +2149,9 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci,
{
int err, rc;
unsigned int i;
- unsigned ioport_size, iomem_size, iomem2_size;
+ unsigned iomem_size;
+ __maybe_unused unsigned ioport_size;
+ __maybe_unused unsigned iomem2_size;
struct snd_korg1212 * korg1212;
const struct firmware *dsp_code;
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 7ba487443c7f..efff220b26ea 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr,
case PIPE_RUNNING:
if(rate != 0)
break;
- /* fall through */
+ fallthrough;
default:
if(rate == 0)
return 0; /* nothing to do */
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index 048a2660d18d..0bdd33b0af65 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id)
dev_err(&mgr->pci->dev,
"canceled notification %x !\n", msg);
}
- /* fall through */
+ fallthrough;
case MSG_TYPE_ANSWER:
/* answer or notification to a message we are waiting for*/
mutex_lock(&mgr->msg_lock);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index ebce4d259e06..975994623c2c 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd)
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
s->suspended = 0;
- /* fallthru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_START:
if (! s->running) {
snd_nm256_playback_start(chip, s, substream);
@@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd)
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
s->suspended = 1;
- /* fallthru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_STOP:
if (s->running) {
snd_nm256_playback_stop(chip);
@@ -1632,11 +1632,11 @@ __error:
}
-enum { NM_BLACKLISTED, NM_RESET_WORKAROUND, NM_RESET_WORKAROUND_2 };
+enum { NM_IGNORED, NM_RESET_WORKAROUND, NM_RESET_WORKAROUND_2 };
static const struct snd_pci_quirk nm256_quirks[] = {
/* HP omnibook 4150 has cs4232 codec internally */
- SND_PCI_QUIRK(0x103c, 0x0007, "HP omnibook 4150", NM_BLACKLISTED),
+ SND_PCI_QUIRK(0x103c, 0x0007, "HP omnibook 4150", NM_IGNORED),
/* Reset workarounds to avoid lock-ups */
SND_PCI_QUIRK(0x104d, 0x8041, "Sony PCG-F305", NM_RESET_WORKAROUND),
SND_PCI_QUIRK(0x1028, 0x0080, "Dell Latitude LS", NM_RESET_WORKAROUND),
@@ -1658,13 +1658,13 @@ static int snd_nm256_probe(struct pci_dev *pci,
dev_dbg(&pci->dev, "Enabled quirk for %s.\n",
snd_pci_quirk_name(q));
switch (q->value) {
- case NM_BLACKLISTED:
+ case NM_IGNORED:
dev_info(&pci->dev,
- "The device is blacklisted. Loading stopped\n");
+ "The device is on the denylist. Loading stopped\n");
return -ENODEV;
case NM_RESET_WORKAROUND_2:
reset_workaround_2 = 1;
- /* Fall-through */
+ fallthrough;
case NM_RESET_WORKAROUND:
reset_workaround = 1;
break;
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index 75b25ecf83a9..b2a3fcfe31d4 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -137,7 +137,7 @@ static int oxygen_open(struct snd_pcm_substream *substream,
SNDRV_PCM_RATE_64000);
runtime->hw.rate_min = 44100;
}
- /* fall through */
+ fallthrough;
case PCM_A:
case PCM_B:
runtime->hw.fifo_size = 0;
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index c3f8721624cd..b90421a1d909 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -29,7 +29,7 @@
* GPIO 4 <- headphone detect
* GPIO 5 -> enable ADC analog circuit for the left channel
* GPIO 6 -> enable ADC analog circuit for the right channel
- * GPIO 7 -> switch green rear output jack between CS4245 and and the first
+ * GPIO 7 -> switch green rear output jack between CS4245 and the first
* channel of CS4361 (mechanical relay)
* GPIO 8 -> enable output to speakers
*
diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c
index 0767276582ca..8aa92f3e5ee8 100644
--- a/sound/pci/oxygen/xonar_wm87x6.c
+++ b/sound/pci/oxygen/xonar_wm87x6.c
@@ -116,7 +116,8 @@ static void wm8776_write(struct oxygen *chip,
else
wm8776_write_i2c(chip, reg, value);
if (reg < ARRAY_SIZE(data->wm8776_regs)) {
- if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER)
+ /* reg >= WM8776_HPLVOL is always true */
+ if (reg <= WM8776_DACMASTER)
value &= ~WM8776_UPDATE;
data->wm8776_regs[reg] = value;
}
@@ -144,7 +145,8 @@ static void wm8766_write(struct oxygen *chip,
OXYGEN_SPI_CEN_LATCH_CLOCK_LO,
(reg << 9) | value);
if (reg < ARRAY_SIZE(data->wm8766_regs)) {
- if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) ||
+ /* reg >= WM8766_LDA1 is always true */
+ if (reg <= WM8766_RDA1 ||
(reg >= WM8766_LDA2 && reg <= WM8766_MASTDA))
value &= ~WM8766_UPDATE;
data->wm8766_regs[reg] = value;
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index f7cda9062088..0fa49f4d15cf 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3027,8 +3027,8 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
- if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
- (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
+ /* syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD is always true */
+ if (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN) {
return syncref;
}
return HDSPM_AES32_AUTOSYNC_FROM_NONE;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 8b03e2dc503f..154d88ce8813 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -544,7 +544,7 @@ static int snd_via82xx_codec_valid(struct via82xx *chip, int secondary)
static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
{
struct via82xx *chip = ac97->private_data;
- int err;
+ __always_unused int err;
err = snd_via82xx_codec_ready(chip, ac97->num);
/* here we need to wait fairly for long time.. */
if (!nodelay)
@@ -2419,7 +2419,7 @@ static const struct via823x_info via823x_cards[] = {
* auto detection of DXS channel supports.
*/
-static const struct snd_pci_quirk dxs_whitelist[] = {
+static const struct snd_pci_quirk dxs_allowlist[] = {
SND_PCI_QUIRK(0x1005, 0x4710, "Avance Logic Mobo", VIA_DXS_ENABLE),
SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K),
SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA),
@@ -2467,9 +2467,9 @@ static int check_dxs_list(struct pci_dev *pci, int revision)
{
const struct snd_pci_quirk *w;
- w = snd_pci_quirk_lookup(pci, dxs_whitelist);
+ w = snd_pci_quirk_lookup(pci, dxs_allowlist);
if (w) {
- dev_dbg(&pci->dev, "DXS white list for %s found\n",
+ dev_dbg(&pci->dev, "DXS allow list for %s found\n",
snd_pci_quirk_name(w));
return w->value;
}
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 607b7100db1c..addfa196df21 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -398,7 +398,7 @@ static int snd_via82xx_codec_valid(struct via82xx_modem *chip, int secondary)
static void snd_via82xx_codec_wait(struct snd_ac97 *ac97)
{
struct via82xx_modem *chip = ac97->private_data;
- int err;
+ __always_unused int err;
err = snd_via82xx_codec_ready(chip, ac97->num);
/* here we need to wait fairly for long time.. */
msleep(500);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index db7d76a3cfeb..cacc6a9d14c8 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream,
kctl = chip->pcm_mixer[substream->number].ctl;
kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
}
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 6c33ea91cc05..27d9da6d61e8 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -139,6 +139,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
/**
* snd_pdacf_assign_resources - initialize the hardware and card instance.
+ * @pdacf: context
* @port: i/o port for the card
* @irq: irq number for the card
*
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
index 6cbe5cb34358..dfc4295b69c4 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c
@@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd)
case SNDRV_PCM_TRIGGER_START:
chip->pcm_hwptr = 0;
chip->pcm_tdone = 0;
- /* fall thru */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
mask = 0;
@@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs)
case SNDRV_PCM_FORMAT_S24_3LE:
case SNDRV_PCM_FORMAT_S24_3BE:
chip->pcm_sample = 3;
- /* fall through */
+ fallthrough;
default: /* 24-bit */
aval = AK4117_DIF_24R;
chip->pcm_frame = 3;
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 73c0fd7277e6..53d558b2806c 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -1063,12 +1063,12 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
if (pm5500 || imac || lombard) {
vmaster_sw = snd_ctl_make_virtual_master(
"Master Playback Switch", (unsigned int *) NULL);
- err = snd_ctl_add_slave_uncached(vmaster_sw,
- chip->master_sw_ctl);
+ err = snd_ctl_add_follower_uncached(vmaster_sw,
+ chip->master_sw_ctl);
if (err < 0)
return err;
- err = snd_ctl_add_slave_uncached(vmaster_sw,
- chip->speaker_sw_ctl);
+ err = snd_ctl_add_follower_uncached(vmaster_sw,
+ chip->speaker_sw_ctl);
if (err < 0)
return err;
err = snd_ctl_add(chip->card, vmaster_sw);
@@ -1076,10 +1076,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
return err;
vmaster_vol = snd_ctl_make_virtual_master(
"Master Playback Volume", (unsigned int *) NULL);
- err = snd_ctl_add_slave(vmaster_vol, master_vol);
+ err = snd_ctl_add_follower(vmaster_vol, master_vol);
if (err < 0)
return err;
- err = snd_ctl_add_slave(vmaster_vol, speaker_vol);
+ err = snd_ctl_add_follower(vmaster_vol, speaker_vol);
if (err < 0)
return err;
err = snd_ctl_add(chip->card, vmaster_vol);
diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c
index f23956cf4ed8..28f039adfa13 100644
--- a/sound/soc/codecs/cros_ec_codec.c
+++ b/sound/soc/codecs/cros_ec_codec.c
@@ -103,28 +103,6 @@ error:
return ret;
}
-static int calculate_sha256(struct cros_ec_codec_priv *priv,
- uint8_t *buf, uint32_t size, uint8_t *digest)
-{
- struct sha256_state sctx;
-
- sha256_init(&sctx);
- sha256_update(&sctx, buf, size);
- sha256_final(&sctx, digest);
-
-#ifdef DEBUG
- {
- char digest_str[65];
-
- bin2hex(digest_str, digest, 32);
- digest_str[64] = 0;
- dev_dbg(priv->dev, "hash=%s\n", digest_str);
- }
-#endif
-
- return 0;
-}
-
static int dmic_get_gain(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -782,9 +760,8 @@ static int wov_hotword_model_put(struct snd_kcontrol *kcontrol,
if (IS_ERR(buf))
return PTR_ERR(buf);
- ret = calculate_sha256(priv, buf, size, digest);
- if (ret)
- goto leave;
+ sha256(buf, size, digest);
+ dev_dbg(priv->dev, "hash=%*phN\n", SHA256_DIGEST_SIZE, digest);
p.cmd = EC_CODEC_WOV_GET_LANG;
ret = send_ec_host_command(priv->ec_device, EC_CMD_EC_CODEC_WOV,
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index a22562f2df47..cdff739924e2 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -680,10 +680,11 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
- FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
- FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+ FSL_SAI_MAXBURST_RX - 1);
snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
&sai->dma_params_rx);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 76b15deea80c..6aba7d28f5f3 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -94,7 +94,7 @@
#define FSL_SAI_CSR_FRDE BIT(0)
/* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK 0x1f
+#define FSL_SAI_CR1_RFW_MASK(x) ((x) - 1)
/* SAI Transmit and Receive Configuration 2 Register */
#define FSL_SAI_CR2_SYNC BIT(30)
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 5176be165210..2b77010c2c5c 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -327,20 +327,22 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np,
return ret;
if (axg_card_cpu_is_playback_fe(dai_link->cpus->of_node))
- ret = meson_card_set_fe_link(card, dai_link, np, true);
+ return meson_card_set_fe_link(card, dai_link, np, true);
else if (axg_card_cpu_is_capture_fe(dai_link->cpus->of_node))
- ret = meson_card_set_fe_link(card, dai_link, np, false);
- else
- ret = meson_card_set_be_link(card, dai_link, np);
+ return meson_card_set_fe_link(card, dai_link, np, false);
+
+ ret = meson_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
- if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
- ret = axg_card_parse_tdm(card, np, index);
- else if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
+ if (axg_card_cpu_is_codec(dai_link->cpus->of_node)) {
dai_link->params = &codec_params;
- dai_link->no_pcm = 0; /* link is not a DPCM BE */
+ } else {
+ dai_link->no_pcm = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+ if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node))
+ ret = axg_card_parse_tdm(card, np, index);
}
return ret;
diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c
index 6da8535f4dd2..5119434a81c4 100644
--- a/sound/soc/meson/gx-card.c
+++ b/sound/soc/meson/gx-card.c
@@ -96,21 +96,21 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np,
return ret;
if (gx_card_cpu_identify(dai_link->cpus, "FIFO"))
- ret = meson_card_set_fe_link(card, dai_link, np, true);
- else
- ret = meson_card_set_be_link(card, dai_link, np);
+ return meson_card_set_fe_link(card, dai_link, np, true);
+ ret = meson_card_set_be_link(card, dai_link, np);
if (ret)
return ret;
- /* Check if the cpu is the i2s encoder and parse i2s data */
- if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
- ret = gx_card_parse_i2s(card, np, index);
-
/* Or apply codec to codec params if necessary */
- else if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
+ if (gx_card_cpu_identify(dai_link->cpus, "CODEC CTRL")) {
dai_link->params = &codec_params;
- dai_link->no_pcm = 0; /* link is not a DPCM BE */
+ } else {
+ dai_link->no_pcm = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+ /* Check if the cpu is the i2s encoder and parse i2s data */
+ if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder"))
+ ret = gx_card_parse_i2s(card, np, index);
}
return ret;
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index f9ce03f3921f..6a64ac01b5ca 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -147,10 +147,6 @@ int meson_card_set_be_link(struct snd_soc_card *card,
struct device_node *np;
int ret, num_codecs;
- link->no_pcm = 1;
- link->dpcm_playback = 1;
- link->dpcm_capture = 1;
-
num_codecs = of_get_child_count(node);
if (!num_codecs) {
dev_err(card->dev, "be link %s has no codec\n",
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe23e936e2d1..2fe1b2ec7c8f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -446,7 +446,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
dev->parent = card->dev;
dev->release = soc_release_rtd_dev;
- dev->groups = soc_dev_attr_groups;
dev_set_name(dev, "%s", dai_link->name);
@@ -503,6 +502,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
/* see for_each_card_rtds */
list_add_tail(&rtd->list, &card->rtd_list);
+ ret = device_add_groups(dev, soc_dev_attr_groups);
+ if (ret < 0)
+ goto free_rtd;
+
return rtd;
free_rtd:
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 693893420bf0..91a2551e4cef 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -402,28 +402,30 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
struct snd_soc_dai_link_component *codec;
struct snd_soc_dai *dai;
bool supported[SNDRV_PCM_STREAM_LAST + 1];
+ bool supported_cpu;
+ bool supported_codec;
int direction;
int i;
for_each_pcm_streams(direction) {
- supported[direction] = true;
+ supported_cpu = false;
+ supported_codec = false;
for_each_link_cpus(dai_link, i, cpu) {
dai = snd_soc_find_dai(cpu);
- if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
- supported[direction] = false;
+ if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+ supported_cpu = true;
break;
}
}
- if (!supported[direction])
- continue;
for_each_link_codecs(dai_link, i, codec) {
dai = snd_soc_find_dai(codec);
- if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
- supported[direction] = false;
+ if (dai && snd_soc_dai_stream_valid(dai, direction)) {
+ supported_codec = true;
break;
}
}
+ supported[direction] = supported_cpu && supported_codec;
}
dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 10f703986be3..00ac1cbf6f88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2743,30 +2743,36 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
if (rtd->dai_link->dpcm_playback) {
stream = SNDRV_PCM_STREAM_PLAYBACK;
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- if (!snd_soc_dai_stream_valid(cpu_dai,
- stream)) {
- dev_err(rtd->card->dev,
- "CPU DAI %s for rtd %s does not support playback\n",
- cpu_dai->name,
- rtd->dai_link->stream_name);
- return -EINVAL;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+ playback = 1;
+ break;
}
- playback = 1;
+ }
+
+ if (!playback) {
+ dev_err(rtd->card->dev,
+ "No CPU DAIs support playback for stream %s\n",
+ rtd->dai_link->stream_name);
+ return -EINVAL;
+ }
}
if (rtd->dai_link->dpcm_capture) {
stream = SNDRV_PCM_STREAM_CAPTURE;
- for_each_rtd_cpu_dais(rtd, i, cpu_dai)
- if (!snd_soc_dai_stream_valid(cpu_dai,
- stream)) {
- dev_err(rtd->card->dev,
- "CPU DAI %s for rtd %s does not support capture\n",
- cpu_dai->name,
- rtd->dai_link->stream_name);
- return -EINVAL;
+ for_each_rtd_cpu_dais(rtd, i, cpu_dai) {
+ if (snd_soc_dai_stream_valid(cpu_dai, stream)) {
+ capture = 1;
+ break;
}
- capture = 1;
+ }
+
+ if (!capture) {
+ dev_err(rtd->card->dev,
+ "No CPU DAIs support capture for stream %s\n",
+ rtd->dai_link->stream_name);
+ return -EINVAL;
+ }
}
} else {
/* Adapt stream for codec2codec links */
diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h
index b04b728c7224..5e159ab239fa 100644
--- a/sound/soc/sof/probe.h
+++ b/sound/soc/sof/probe.h
@@ -36,7 +36,7 @@ struct sof_probe_point_desc {
struct sof_ipc_probe_dma_add_params {
struct sof_ipc_cmd_hdr hdr;
unsigned int num_elems;
- struct sof_probe_dma dma[0];
+ struct sof_probe_dma dma[];
} __packed;
struct sof_ipc_probe_info_params {
@@ -51,19 +51,19 @@ struct sof_ipc_probe_info_params {
struct sof_ipc_probe_dma_remove_params {
struct sof_ipc_cmd_hdr hdr;
unsigned int num_elems;
- unsigned int stream_tag[0];
+ unsigned int stream_tag[];
} __packed;
struct sof_ipc_probe_point_add_params {
struct sof_ipc_cmd_hdr hdr;
unsigned int num_elems;
- struct sof_probe_point_desc desc[0];
+ struct sof_probe_point_desc desc[];
} __packed;
struct sof_ipc_probe_point_remove_params {
struct sof_ipc_cmd_hdr hdr;
unsigned int num_elems;
- unsigned int buffer_id[0];
+ unsigned int buffer_id[];
} __packed;
int sof_ipc_probe_init(struct snd_sof_dev *sdev,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index cf7049999261..913adc8568d5 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -22,7 +22,7 @@
* - Data sheet of the T7903, a newer but very similar ISA bus equivalent
* available from the Lucent (formerly AT&T microelectronics) home
* page.
- * - http://www.freesoft.org/Linux/DBRI/
+ * - https://www.freesoft.org/Linux/DBRI/
* - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec
* Interfaces: CHI, Audio In & Out, 2 bits parallel
* Documentation: from the Crystal Semiconductor home page.
@@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len)
switch (len) {
case 32:
b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16);
- /* fall through */
+ fallthrough;
case 16:
b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8);
- /* fall through */
+ fallthrough;
case 8:
b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4);
- /* fall through */
+ fallthrough;
case 4:
b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2);
- /* fall through */
+ fallthrough;
case 2:
b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1);
case 1:
diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c
index 20f34d2ace5f..9bd8dcbb68e4 100644
--- a/sound/usb/6fire/control.c
+++ b/sound/usb/6fire/control.c
@@ -539,7 +539,7 @@ static int usb6fire_control_add_virtual(
ret = snd_ctl_add(card, control);
if (ret < 0)
return ret;
- ret = snd_ctl_add_slave(vmaster, control);
+ ret = snd_ctl_add_follower(vmaster, control);
if (ret < 0)
return ret;
i++;
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index e9243d53a107..3b6bb2cbe886 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -820,7 +820,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *cdev)
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
cdev->samplerates |= SNDRV_PCM_RATE_192000;
- /* fall thru */
+ fallthrough;
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO2DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index b669e119f654..2af3b7eb0a88 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -187,7 +187,7 @@ static void usb_ep1_command_reply_dispatch (struct urb* urb)
break;
}
#ifdef CONFIG_SND_USB_CAIAQ_INPUT
- /* fall through */
+ fallthrough;
case EP1_CMD_READ_ERP:
case EP1_CMD_READ_ANALOG:
snd_usb_caiaq_input_dispatch(cdev, buf, urb->actual_length);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 162bdd6eb4d4..696e788c5d31 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
dev_warn(&dev->dev,
"unknown interface protocol %#02x, assuming v1\n",
protocol);
- /* fall through */
+ fallthrough;
case UAC_VERSION_1: {
struct uac1_ac_header_descriptor *h1;
diff --git a/sound/usb/card.h b/sound/usb/card.h
index d6219fba9699..5351d7183b1b 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -84,10 +84,10 @@ struct snd_usb_endpoint {
dma_addr_t sync_dma; /* DMA address of syncbuf */
unsigned int pipe; /* the data i/o pipe */
- unsigned int framesize[2]; /* small/large frame sizes in samples */
- unsigned int sample_rem; /* remainder from division fs/fps */
+ unsigned int packsize[2]; /* small/large packet sizes in samples */
+ unsigned int sample_rem; /* remainder from division fs/pps */
unsigned int sample_accum; /* sample accumulator */
- unsigned int fps; /* frames per second */
+ unsigned int pps; /* packets per second */
unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */
unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */
int freqshift; /* how much to shift the feedback value to get Q16.16 */
@@ -137,6 +137,7 @@ struct snd_usb_substream {
unsigned int tx_length_quirk:1; /* add length specifier to transfers */
unsigned int fmt_type; /* USB audio format type (1-3) */
unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */
+ unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */
unsigned int running: 1; /* running status */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b118cf97607f..f3ca59005d91 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
else
return 0;
}
- /* fall through */
+ fallthrough;
case UAC_VERSION_2:
return set_sample_rate_v2v3(chip, iface, alts, fmt, rate);
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 9bea7d3f99f8..5fbc8dd2f409 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -159,11 +159,11 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
return ep->maxframesize;
ep->sample_accum += ep->sample_rem;
- if (ep->sample_accum >= ep->fps) {
- ep->sample_accum -= ep->fps;
- ret = ep->framesize[1];
+ if (ep->sample_accum >= ep->pps) {
+ ep->sample_accum -= ep->pps;
+ ret = ep->packsize[1];
} else {
- ret = ep->framesize[0];
+ ret = ep->packsize[0];
}
return ret;
@@ -335,7 +335,7 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
while (test_bit(EP_FLAG_RUNNING, &ep->flags)) {
unsigned long flags;
- struct snd_usb_packet_info *uninitialized_var(packet);
+ struct snd_usb_packet_info *packet;
struct snd_urb_ctx *ctx = NULL;
int err, i;
@@ -615,9 +615,8 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force)
for (i = 0; i < ep->nurbs; i++)
release_urb_ctx(&ep->urb[i]);
- if (ep->syncbuf)
- usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
- ep->syncbuf, ep->sync_dma);
+ usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
+ ep->syncbuf, ep->sync_dma);
ep->syncbuf = NULL;
ep->nurbs = 0;
@@ -1088,15 +1087,15 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
ep->freqn = get_usb_full_speed_rate(rate);
- ep->fps = 1000;
+ ep->pps = 1000 >> ep->datainterval;
} else {
ep->freqn = get_usb_high_speed_rate(rate);
- ep->fps = 8000;
+ ep->pps = 8000 >> ep->datainterval;
}
- ep->sample_rem = rate % ep->fps;
- ep->framesize[0] = rate / ep->fps;
- ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
+ ep->sample_rem = rate % ep->pps;
+ ep->packsize[0] = rate / ep->pps;
+ ep->packsize[1] = (rate + (ep->pps - 1)) / ep->pps;
/* calculate the frequency in 16.16 format */
ep->freqm = ep->freqn;
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 5ffb457cc88c..1b28d01d1f4c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -394,8 +394,9 @@ skip_rate:
return nr_rates;
}
-/* Line6 Helix series don't support the UAC2_CS_RANGE usb function
- * call. Return a static table of known clock rates.
+/* Line6 Helix series and the Rode Rodecaster Pro don't support the
+ * UAC2_CS_RANGE usb function call. Return a static table of known
+ * clock rates.
*/
static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
struct audioformat *fp)
@@ -408,6 +409,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip,
case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */
case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */
case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */
+ case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */
return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000);
}
diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c
index 663d608c4287..970c9bdce0b2 100644
--- a/sound/usb/line6/capture.c
+++ b/sound/usb/line6/capture.c
@@ -286,6 +286,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm)
urb->interval = LINE6_ISO_INTERVAL;
urb->error_count = 0;
urb->complete = audio_in_callback;
+ if (usb_urb_ep_type_check(urb))
+ return -EINVAL;
}
return 0;
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 7629116f570e..60674ce4879b 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6)
/*
Send raw message in pieces of wMaxPacketSize bytes.
*/
-static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
+int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
int size)
{
int i, done = 0;
@@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer,
return done;
}
+EXPORT_SYMBOL_GPL(line6_send_raw_message);
/*
Notification of completion of asynchronous request transmission.
@@ -840,7 +841,7 @@ void line6_disconnect(struct usb_interface *interface)
if (WARN_ON(usbdev != line6->usbdev))
return;
- cancel_delayed_work(&line6->startup_work);
+ cancel_delayed_work_sync(&line6->startup_work);
if (line6->urb_listen != NULL)
line6_stop_listen(line6);
diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h
index 1a4e3700c80c..71d3da1db8c8 100644
--- a/sound/usb/line6/driver.h
+++ b/sound/usb/line6/driver.h
@@ -66,8 +66,8 @@
extern const unsigned char line6_midi_id[3];
-static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3;
-static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4;
+#define SYSEX_DATA_OFS (sizeof(line6_midi_id) + 3)
+#define SYSEX_EXTRA_SIZE (sizeof(line6_midi_id) + 4)
/*
Common properties of Line 6 devices.
@@ -108,6 +108,8 @@ enum {
LINE6_CAP_CONTROL_MIDI = 1 << 4,
/* device provides low-level information */
LINE6_CAP_CONTROL_INFO = 1 << 5,
+ /* device provides hardware monitoring volume control */
+ LINE6_CAP_HWMON_CTL = 1 << 6,
};
/*
@@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address,
void *data, unsigned datalen);
extern int line6_read_serial_number(struct usb_line6 *line6,
u32 *serial_number);
+extern int line6_send_raw_message(struct usb_line6 *line6,
+ const char *buffer, int size);
extern int line6_send_raw_message_async(struct usb_line6 *line6,
const char *buffer, int size);
extern int line6_send_sysex_message(struct usb_line6 *line6,
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 01930ce7bd75..8233c61e23f1 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -431,6 +431,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm)
urb->interval = LINE6_ISO_INTERVAL;
urb->error_count = 0;
urb->complete = audio_out_callback;
+ if (usb_urb_ep_type_check(urb))
+ return -EINVAL;
}
return 0;
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index e39dc85c355a..eef45f7fef0d 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -11,6 +11,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/core.h>
+#include <sound/control.h>
#include <sound/pcm.h>
#include "driver.h"
@@ -37,6 +38,9 @@ struct usb_line6_podhd {
/* Firmware version */
int firmware_version;
+
+ /* Monitor level */
+ int monitor_level;
};
#define line6_to_podhd(x) container_of(x, struct usb_line6_podhd, line6)
@@ -250,6 +254,116 @@ static void podhd_disconnect(struct usb_line6 *line6)
}
}
+static const unsigned int float_zero_to_one_lookup[] = {
+0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd,
+0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae,
+0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b,
+0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f,
+0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1,
+0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333,
+0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85,
+0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7,
+0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14,
+0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d,
+0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666,
+0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f,
+0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8,
+0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1,
+0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a,
+0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333,
+0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000
+};
+
+static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value)
+{
+ unsigned int fl;
+ static const unsigned char msg[16] = {
+ /* Chunk is 0xc bytes (without first word) */
+ 0x0c, 0x00,
+ /* First chunk in the message */
+ 0x01, 0x00,
+ /* Message size is 2 4-byte words */
+ 0x02, 0x00,
+ /* Unknown */
+ 0x04, 0x41,
+ /* Unknown */
+ 0x04, 0x00, 0x13, 0x00,
+ /* Volume, LE float32, 0.0 - 1.0 */
+ 0x00, 0x00, 0x00, 0x00
+ };
+ unsigned char *buf;
+
+ buf = kmemdup(msg, sizeof(msg), GFP_KERNEL);
+ if (!buf)
+ return;
+
+ if (value < 0)
+ value = 0;
+
+ if (value >= ARRAY_SIZE(float_zero_to_one_lookup))
+ value = ARRAY_SIZE(float_zero_to_one_lookup) - 1;
+
+ fl = float_zero_to_one_lookup[value];
+
+ buf[12] = (fl >> 0) & 0xff;
+ buf[13] = (fl >> 8) & 0xff;
+ buf[14] = (fl >> 16) & 0xff;
+ buf[15] = (fl >> 24) & 0xff;
+
+ line6_send_raw_message(&podhd->line6, buf, sizeof(msg));
+ kfree(buf);
+
+ podhd->monitor_level = value;
+}
+
+/* control info callback */
+static int snd_podhd_control_monitor_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+/* control get callback */
+static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol);
+ struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6);
+
+ ucontrol->value.integer.value[0] = podhd->monitor_level;
+ return 0;
+}
+
+/* control put callback */
+static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol);
+ struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6);
+
+ if (ucontrol->value.integer.value[0] == podhd->monitor_level)
+ return 0;
+
+ podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]);
+ return 1;
+}
+
+/* control definition */
+static const struct snd_kcontrol_new podhd_control_monitor = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Monitor Playback Volume",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_podhd_control_monitor_info,
+ .get = snd_podhd_control_monitor_get,
+ .put = snd_podhd_control_monitor_put
+};
+
/*
Try to init POD HD device.
*/
@@ -298,6 +412,15 @@ static int podhd_init(struct usb_line6 *line6,
return err;
}
+ if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) {
+ podhd_set_monitor_level(pod, 100);
+ err = snd_ctl_add(line6->card,
+ snd_ctl_new1(&podhd_control_monitor,
+ line6->line6pcm));
+ if (err < 0)
+ return err;
+ }
+
if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) {
/* register USB audio system directly */
return snd_card_register(line6->card);
@@ -354,7 +477,7 @@ static const struct line6_properties podhd_properties_table[] = {
.id = "PODHD500",
.name = "POD HD500",
.capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL
- | LINE6_CAP_HWMON,
+ | LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL,
.altsetting = 1,
.ctrl_if = 1,
.ep_ctrl_r = 0x81,
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 047b90595d65..df639fe03118 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1499,6 +1499,8 @@ void snd_usbmidi_disconnect(struct list_head *p)
spin_unlock_irq(&umidi->disc_lock);
up_write(&umidi->disc_rwsem);
+ del_timer_sync(&umidi->error_timer);
+
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
if (ep->out)
@@ -1525,7 +1527,6 @@ void snd_usbmidi_disconnect(struct list_head *p)
ep->in = NULL;
}
}
- del_timer_sync(&umidi->error_timer);
}
EXPORT_SYMBOL(snd_usbmidi_disconnect);
@@ -2301,16 +2302,22 @@ void snd_usbmidi_input_stop(struct list_head *p)
}
EXPORT_SYMBOL(snd_usbmidi_input_stop);
-static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep)
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_in_endpoint *ep)
{
unsigned int i;
+ unsigned long flags;
if (!ep)
return;
for (i = 0; i < INPUT_URBS; ++i) {
struct urb *urb = ep->urbs[i];
- urb->dev = ep->umidi->dev;
- snd_usbmidi_submit_urb(urb, GFP_KERNEL);
+ spin_lock_irqsave(&umidi->disc_lock, flags);
+ if (!atomic_read(&urb->use_count)) {
+ urb->dev = ep->umidi->dev;
+ snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
+ }
+ spin_unlock_irqrestore(&umidi->disc_lock, flags);
}
}
@@ -2326,7 +2333,7 @@ void snd_usbmidi_input_start(struct list_head *p)
if (umidi->input_running || !umidi->opened[1])
return;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
- snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+ snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in);
umidi->input_running = 1;
}
EXPORT_SYMBOL(snd_usbmidi_input_start);
@@ -2401,7 +2408,7 @@ int __snd_usbmidi_create(struct snd_card *card,
break;
case QUIRK_MIDI_US122L:
umidi->usb_protocol_ops = &snd_usbmidi_122l_ops;
- /* fall through */
+ fallthrough;
case QUIRK_MIDI_FIXED_ENDPOINT:
memcpy(&endpoints[0], quirk->data,
sizeof(struct snd_usb_midi_endpoint_info));
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 15769f266790..81e987eaf063 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -581,8 +581,9 @@ static int check_matrix_bitmap(unsigned char *bmap,
* if failed, give up and free the control instance.
*/
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
- struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+ struct snd_kcontrol *kctl,
+ bool is_std_info)
{
struct usb_mixer_interface *mixer = list->mixer;
int err;
@@ -596,6 +597,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
return err;
}
list->kctl = kctl;
+ list->is_std_info = is_std_info;
list->next_id_elem = mixer->id_elems[list->id];
mixer->id_elems[list->id] = list;
return 0;
@@ -1461,6 +1463,10 @@ static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol,
snd_usb_unlock_shutdown(chip);
if (ret < 0) {
+ if (strstr(kcontrol->id.name, "Speaker")) {
+ ucontrol->value.integer.value[0] = 1;
+ return 0;
+ }
error:
usb_audio_err(chip,
"cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
@@ -2365,7 +2371,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
int num_ins;
struct usb_mixer_elem_info *cval;
struct snd_kcontrol *kctl;
- int i, err, nameid, type, len;
+ int i, err, nameid, type, len, val;
const struct procunit_info *info;
const struct procunit_value_info *valinfo;
const struct usbmix_name_map *map;
@@ -2468,6 +2474,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
break;
}
+ err = get_cur_ctl_value(cval, cval->control << 8, &val);
+ if (err < 0) {
+ usb_mixer_elem_info_free(cval);
+ return -EINVAL;
+ }
+
kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
if (!kctl) {
usb_mixer_elem_info_free(cval);
@@ -3234,8 +3246,11 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
unitid = delegate_notify(mixer, unitid, NULL, NULL);
for_each_mixer_elem(list, mixer, unitid) {
- struct usb_mixer_elem_info *info =
- mixer_elem_list_to_info(list);
+ struct usb_mixer_elem_info *info;
+
+ if (!list->is_std_info)
+ continue;
+ info = mixer_elem_list_to_info(list);
/* invalidate cache, so the value is read from the device */
info->cached = 0;
snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -3315,6 +3330,8 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
if (!list->kctl)
continue;
+ if (!list->is_std_info)
+ continue;
info = mixer_elem_list_to_info(list);
if (count > 1 && info->control != control)
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 41ec9dc4139b..c29e27ac43a7 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -66,6 +66,7 @@ struct usb_mixer_elem_list {
struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */
struct snd_kcontrol *kctl;
unsigned int id;
+ bool is_std_info;
usb_mixer_elem_dump_func_t dump;
usb_mixer_elem_resume_func_t resume;
};
@@ -103,8 +104,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid);
int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
int request, int validx, int value_set);
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
- struct snd_kcontrol *kctl);
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+ struct snd_kcontrol *kctl,
+ bool is_std_info);
+
+#define snd_usb_mixer_add_control(list, kctl) \
+ snd_usb_mixer_add_list(list, kctl, true)
void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
struct usb_mixer_interface *mixer,
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 9af7aa93f6fa..5b43e9e40e49 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -233,7 +233,7 @@ static const struct usbmix_name_map maya44_map[] = {
};
/* Section "justlink_map" below added by James Courtier-Dutton <James@superbug.demon.co.uk>
- * sourced from Maplin Electronics (http://www.maplin.co.uk), part number A56AK
+ * sourced from Maplin Electronics (https://www.maplin.co.uk), part number A56AK
* Part has 2 connectors that act as a single output. (TOSLINK Optical for digital out, and 3.5mm Jack for Analogue out.)
* The USB Mixer publishes a Microphone and extra Volume controls for it, but none exist on the device,
* so this map removes all unwanted sliders from alsamixer
@@ -370,6 +370,12 @@ static const struct usbmix_name_map asus_rog_map[] = {
{}
};
+static const struct usbmix_name_map lenovo_p620_rear_map[] = {
+ { 19, NULL, 2 }, /* FU, Volume */
+ { 19, NULL, 12 }, /* FU, Input Gain Pad */
+ {}
+};
+
/* TRX40 mobos with Realtek ALC1220-VB */
static const struct usbmix_name_map trx40_mobo_map[] = {
{ 18, NULL }, /* OT, IEC958 - broken response, disabled */
@@ -573,6 +579,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = trx40_mobo_map,
.connector_map = trx40_mobo_connector_map,
},
+ { /* Lenovo ThinkStation P620 Rear */
+ .id = USB_ID(0x17aa, 0x1046),
+ .map = lenovo_p620_rear_map,
+ },
{ 0 } /* terminator */
};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index b6bcf2f92383..199cdbfdc761 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -158,7 +158,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
return -ENOMEM;
}
kctl->private_free = snd_usb_mixer_elem_free;
- return snd_usb_mixer_add_control(list, kctl);
+ /* don't use snd_usb_mixer_add_control() here, this is a special list element */
+ return snd_usb_mixer_add_list(list, kctl, false);
}
/*
@@ -184,6 +185,7 @@ static const struct rc_config {
{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */
{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
{ USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
+ { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */
};
diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c
index 6483e47bafd0..c53a9773f310 100644
--- a/sound/usb/mixer_s1810c.c
+++ b/sound/usb/mixer_s1810c.c
@@ -554,11 +554,11 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer)
dev_info(&dev->dev,
"Presonus Studio 1810c, device_setup: %u\n", chip->setup);
if (chip->setup == 1)
- dev_info(&dev->dev, "(8out/18in @ 48KHz)\n");
+ dev_info(&dev->dev, "(8out/18in @ 48kHz)\n");
else if (chip->setup == 2)
- dev_info(&dev->dev, "(6out/8in @ 192KHz)\n");
+ dev_info(&dev->dev, "(6out/8in @ 192kHz)\n");
else
- dev_info(&dev->dev, "(8out/14in @ 96KHz)\n");
+ dev_info(&dev->dev, "(8out/14in @ 96kHz)\n");
ret = snd_s1810c_init_mixer_maps(chip);
if (ret < 0)
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 74c00c905d24..0ffff7640892 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -332,7 +332,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = {
},
[SCARLETT2_PORT_TYPE_SPDIF] = {
.id = 0x180,
- /* S/PDIF outputs aren't available at 192KHz
+ /* S/PDIF outputs aren't available at 192kHz
* but are included in the USB mux I/O
* assignment message anyway
*/
@@ -401,7 +401,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = {
.dst_descr = "Analogue Output %02d Playback"
},
[SCARLETT2_PORT_TYPE_SPDIF] = {
- /* S/PDIF outputs aren't available at 192KHz
+ /* S/PDIF outputs aren't available at 192kHz
* but are included in the USB mux I/O
* assignment message anyway
*/
diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
index 986145fd2ce0..a4d4d71db55b 100644
--- a/sound/usb/mixer_us16x08.c
+++ b/sound/usb/mixer_us16x08.c
@@ -329,7 +329,7 @@ static int snd_us16x08_bus_put(struct snd_kcontrol *kcontrol,
elem->cached |= 1;
elem->cache_val[0] = val;
} else {
- usb_audio_dbg(chip, "Failed to set buss param, err:%d\n", err);
+ usb_audio_dbg(chip, "Failed to set bus parameter, err:%d\n", err);
}
return err > 0 ? 1 : 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 8a05dcb1344f..5600751803cf 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -367,6 +367,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
ifnum = 0;
goto add_sync_ep_from_ifnum;
case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
+ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
ep = 0x81;
ifnum = 2;
goto add_sync_ep_from_ifnum;
@@ -1417,6 +1420,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs,
// continue;
}
bytes = urb->iso_frame_desc[i].actual_length;
+ if (subs->stream_offset_adj > 0) {
+ unsigned int adj = min(subs->stream_offset_adj, bytes);
+ cp += adj;
+ bytes -= adj;
+ subs->stream_offset_adj -= adj;
+ }
frames = bytes / stride;
if (!subs->txfr_quirk)
bytes = frames * stride;
@@ -1693,8 +1702,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
int processed = urb->transfer_buffer_length / ep->stride;
int est_delay;
- /* ignore the delay accounting when procssed=0 is given, i.e.
- * silent payloads are procssed before handling the actual data
+ /* ignore the delay accounting when processed=0 is given, i.e.
+ * silent payloads are processed before handling the actual data
*/
if (!processed)
return;
@@ -1741,7 +1750,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
subs->trigger_tstamp_pending_update = true;
- /* fall through */
+ fallthrough;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
subs->data_endpoint->retire_data_urb = retire_playback_urb;
@@ -1786,6 +1795,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
return 0;
case SNDRV_PCM_TRIGGER_STOP:
stop_endpoints(subs);
+ subs->data_endpoint->retire_data_urb = NULL;
subs->running = 0;
return 0;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 4ec491011b19..f4fb002e3ef4 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -127,7 +127,7 @@
/*
* HP Wireless Audio
* When not ignored, causes instability issues for some users, forcing them to
- * blacklist the entire module.
+ * skip the entire module.
*/
{
USB_DEVICE(0x0424, 0xb832),
@@ -2680,6 +2680,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 0,
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2690,6 +2694,32 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
.endpoint = 0x01,
.ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x024c,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .nr_rates = 2,
+ .rate_table = (unsigned int[]) {
+ 44100, 48000
+ }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x0126,
.rates = SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000,
.rate_min = 44100,
@@ -2794,6 +2824,19 @@ YAMAHA_DEVICE(0x7010, "UB99"),
QUIRK_RENAME_DEVICE("Rane", "SL-1")
},
+/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
+{
+ USB_DEVICE(0x17aa, 0x1046),
+ QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear",
+ "Lenovo-ThinkStation-P620-Rear"),
+},
+/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
+{
+ USB_DEVICE(0x17aa, 0x104d),
+ QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main",
+ "Lenovo-ThinkStation-P620-Main"),
+},
+
/* Native Instruments MK2 series */
{
/* Komplete Audio 6 */
@@ -3252,11 +3295,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
}
},
+/*
+ * The original product_name is "USB Sound Device", however this name
+ * is also used by the CM106 based cards, so make it unique.
+ */
+{
+ USB_DEVICE(0x0d8c, 0x0102),
+ QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D")
+},
{
- /*
- * The original product_name is "USB Sound Device", however this name
- * is also used by the CM106 based cards, so make it unique.
- */
USB_DEVICE(0x0d8c, 0x0103),
QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII")
},
@@ -3541,6 +3588,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
}
}
},
+{
+ /*
+ * PIONEER DJ DDJ-RB
+ * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed
+ * The feedback for the output is the dummy input.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 4,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x01,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 44100 }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC|
+ USB_ENDPOINT_USAGE_IMPLICIT_FB,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 44100 }
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
#define ALC1220_VB_DESKTOP(vend, prod) { \
USB_DEVICE(vend, prod), \
@@ -3633,4 +3736,62 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
}
},
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
+ */
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x534d,
+ .idProduct = 0x2109,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "MacroSilicon",
+ .product_name = "MS2109",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = &(const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_ALIGN_TRANSFER,
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 2,
+ .iface = 3,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC |
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
#undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index bca0179a0ef8..abf99b814a0f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1144,14 +1144,14 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev)
#define MAUDIO_SET 0x01 /* parse device_setup */
#define MAUDIO_SET_COMPATIBLE 0x80 /* use only "win-compatible" interfaces */
#define MAUDIO_SET_DTS 0x02 /* enable DTS Digital Output */
-#define MAUDIO_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */
+#define MAUDIO_SET_96K 0x04 /* 48-96kHz rate if set, 8-48kHz otherwise */
#define MAUDIO_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */
#define MAUDIO_SET_DI 0x10 /* enable Digital Input */
#define MAUDIO_SET_MASK 0x1f /* bit mask for setup value */
-#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48KHz+Digital Input */
-#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */
-#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48KHz+Digital Input */
-#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */
+#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48kHz+Digital Input */
+#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48kHz+No Digital Input */
+#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48kHz+Digital Input */
+#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48kHz+No Digital Input */
static int quattro_skip_setting_quirk(struct snd_usb_audio *chip,
int iface, int altno)
@@ -1495,6 +1495,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */
pioneer_djm_set_format_quirk(subs);
break;
+ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
+ subs->stream_offset_adj = 2;
+ break;
}
}
@@ -1532,6 +1535,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip)
static bool is_itf_usb_dsd_dac(unsigned int id)
{
switch (id) {
+ case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */
case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
@@ -1596,7 +1600,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep)
/*
* M-Audio Fast Track C400/C600 - when packets are not skipped, real
- * world latency varies by approx. +/- 50 frames (at 96KHz) each time
+ * world latency varies by approx. +/- 50 frames (at 96kHz) each time
* the stream is (re)started. When skipping packets 16 at endpoint
* start up, the real world latency is stable within +/- 1 frame (also
* across power cycles).
@@ -1673,6 +1677,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe,
chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
(requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
usleep_range(1000, 2000);
+
+ /*
+ * Samsung USBC Headset (AKG) need a tiny delay after each
+ * class compliant request. (Model number: AAM625R or AAM627R)
+ */
+ if (chip->usb_id == USB_ID(0x04e8, 0xa051) &&
+ (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+ usleep_range(5000, 6000);
}
/*
@@ -1830,7 +1842,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip,
/*
* MaxPacketsOnly attribute is erroneously set in endpoint
* descriptors. As a result this card produces noise with
- * all sample rates other than 96 KHz.
+ * all sample rates other than 96 kHz.
*/
fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
break;
@@ -1856,6 +1868,7 @@ struct registration_quirk {
static const struct registration_quirk registration_quirks[] = {
REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */
+ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */
{ 0 } /* terminator */
};
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 15296f2c902c..ca76ba5b5c0b 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as,
subs->tx_length_quirk = as->chip->tx_length_quirk;
subs->speed = snd_usb_get_speed(subs->dev);
subs->pkt_offset_adj = 0;
+ subs->stream_offset_adj = 0;
snd_usb_set_pcm_ops(as->pcm, stream);
@@ -1146,9 +1147,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n",
iface_no, altno, protocol);
protocol = UAC_VERSION_1;
- /* fall through */
+ fallthrough;
case UAC_VERSION_1:
- /* fall through */
case UAC_VERSION_2: {
int bm_quirk = 0;
diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c
index a9e5c2cd7698..228d82031297 100644
--- a/sound/xen/xen_snd_front.c
+++ b/sound/xen/xen_snd_front.c
@@ -114,7 +114,7 @@ int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl,
int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl)
{
- struct xensnd_req *req;
+ __always_unused struct xensnd_req *req;
int ret;
mutex_lock(&evtchnl->u.req.req_io_lock);
@@ -246,11 +246,8 @@ static void sndback_changed(struct xenbus_device *xb_dev,
switch (backend_state) {
case XenbusStateReconfiguring:
- /* fall through */
case XenbusStateReconfigured:
- /* fall through */
case XenbusStateInitialised:
- /* fall through */
break;
case XenbusStateInitialising:
@@ -289,7 +286,6 @@ static void sndback_changed(struct xenbus_device *xb_dev,
break;
case XenbusStateUnknown:
- /* fall through */
case XenbusStateClosed:
if (xb_dev->state == XenbusStateClosed)
break;
diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c
index 102d6e096cc8..29e0f0ea67eb 100644
--- a/sound/xen/xen_snd_front_evtchnl.c
+++ b/sound/xen/xen_snd_front_evtchnl.c
@@ -46,13 +46,9 @@ again:
continue;
switch (resp->operation) {
case XENSND_OP_OPEN:
- /* fall through */
case XENSND_OP_CLOSE:
- /* fall through */
case XENSND_OP_READ:
- /* fall through */
case XENSND_OP_WRITE:
- /* fall through */
case XENSND_OP_TRIGGER:
channel->u.req.resp_status = resp->status;
complete(&channel->u.req.completion);