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authorLinus Torvalds <torvalds@linux-foundation.org>2020-08-06 14:27:31 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2020-08-06 14:27:31 -0700
commit3f9df56480fc8ce492fc9e988d67bdea884ed15c (patch)
tree6e1c5ed1e28b72435995b8bcd191daa7dfdf770e /sound/isa
parent921d2597abfc05e303f08baa6ead8f9ab8a723e1 (diff)
parentc7fabbc51352f50cc58242a6dc3b9c1a3599849b (diff)
Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "This became wide and scattered updates all over the sound tree as diffstat shows: lots of (still ongoing) refactoring works in ASoC, fixes and cleanups caught by static analysis, inclusive term conversions as well as lots of new drivers. Below are highlights: ASoC core: - API cleanups and conversions to the unified mute_stream() call - Simplify I/O helper functions - Use helper macros to retrieve RTD from substreams ASoC drivers: - Lots of fixes and cleanups in Intel ASoC drivers - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards, nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards, TI J721e EVM ALSA core: - Minor code refacotring for SG-buffer handling HD-audio: - Generalization of mute-LED handling with LED classdev - Intel silent stream support for HDMI - Device-specific fixes: CA0132, Loongson-3 Others: - Usual USB- and HD-audio quirks for various devices - Fixes for echoaudio DMA position handling - Various documents and trivial fixes for sparse warnings - Conversion to adopt inclusive terms" * tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits) ALSA: pci: delete repeated words in comments ALSA: isa: delete repeated words in comments ALSA: hda/tegra: Add 100us dma stop delay ALSA: hda: Add dma stop delay variable ASoC: hda/tegra: Set buffer alignment to 128 bytes ALSA: seq: oss: Serialize ioctls ALSA: hda/hdmi: Add quirk to force connectivity ALSA: usb-audio: add startech usb audio dock name ALSA: usb-audio: Add support for Lenovo ThinkStation P620 Revert "ALSA: hda: call runtime_allow() for all hda controllers" ALSA: hda/ca0132 - Fix AE-5 microphone selection commands. ALSA: hda/ca0132 - Add new quirk ID for Recon3D. ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value. ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops ALSA: docs: fix typo ALSA: doc: use correct config variable name ASoC: core: Two step component registration ASoC: core: Simplify snd_soc_component_initialize declaration ASoC: core: Relocate and expose snd_soc_component_initialize ASoC: sh: Replace 'select' DMADEVICES 'with depends on' ...
Diffstat (limited to 'sound/isa')
-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/sb8_main.c10
9 files changed, 26 insertions, 27 deletions
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/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)