Linux kernel mirror (for testing)
git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
kernel
os
linux
ASoC: Merge up fixes for CI
Avoid tripping over fixed issues in CI.
+105
-14
+4
-2
Documentation/devicetree/bindings/sound/tas2562.yaml
+4
-2
Documentation/devicetree/bindings/sound/tas2562.yaml
···
55
55
description: TDM TX current sense time slot.
56
56
57
57
'#sound-dai-cells':
58
-
const: 1
58
+
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
59
+
# compatibility but is deprecated.
60
+
enum: [0, 1]
59
61
60
62
required:
61
63
- compatible
···
74
72
codec: codec@4c {
75
73
compatible = "ti,tas2562";
76
74
reg = <0x4c>;
77
-
#sound-dai-cells = <1>;
75
+
#sound-dai-cells = <0>;
78
76
interrupt-parent = <&gpio1>;
79
77
interrupts = <14>;
80
78
shutdown-gpios = <&gpio1 15 0>;
+4
-2
Documentation/devicetree/bindings/sound/tas2770.yaml
+4
-2
Documentation/devicetree/bindings/sound/tas2770.yaml
···
57
57
- 1 # Falling edge
58
58
59
59
'#sound-dai-cells':
60
-
const: 1
60
+
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
61
+
# compatibility but is deprecated.
62
+
enum: [0, 1]
61
63
62
64
required:
63
65
- compatible
···
76
74
codec: codec@41 {
77
75
compatible = "ti,tas2770";
78
76
reg = <0x41>;
79
-
#sound-dai-cells = <1>;
77
+
#sound-dai-cells = <0>;
80
78
interrupt-parent = <&gpio1>;
81
79
interrupts = <14>;
82
80
reset-gpio = <&gpio1 15 0>;
+4
-2
Documentation/devicetree/bindings/sound/tas27xx.yaml
+4
-2
Documentation/devicetree/bindings/sound/tas27xx.yaml
···
50
50
description: TDM TX voltage sense time slot.
51
51
52
52
'#sound-dai-cells':
53
-
const: 1
53
+
# The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward
54
+
# compatibility but is deprecated.
55
+
enum: [0, 1]
54
56
55
57
required:
56
58
- compatible
···
69
67
codec: codec@38 {
70
68
compatible = "ti,tas2764";
71
69
reg = <0x38>;
72
-
#sound-dai-cells = <1>;
70
+
#sound-dai-cells = <0>;
73
71
interrupt-parent = <&gpio1>;
74
72
interrupts = <14>;
75
73
reset-gpios = <&gpio1 15 0>;
+7
sound/soc/amd/yc/acp6x-mach.c
+7
sound/soc/amd/yc/acp6x-mach.c
+15
sound/soc/codecs/ssm2602.c
+15
sound/soc/codecs/ssm2602.c
···
53
53
{ .reg = 0x09, .def = 0x0000 }
54
54
};
55
55
56
+
/*
57
+
* ssm2602 register patch
58
+
* Workaround for playback distortions after power up: activates digital
59
+
* core, and then powers on output, DAC, and whole chip at the same time
60
+
*/
61
+
62
+
static const struct reg_sequence ssm2602_patch[] = {
63
+
{ SSM2602_ACTIVE, 0x01 },
64
+
{ SSM2602_PWR, 0x07 },
65
+
{ SSM2602_RESET, 0x00 },
66
+
};
67
+
56
68
57
69
/*Appending several "None"s just for OSS mixer use*/
58
70
static const char *ssm2602_input_select[] = {
···
609
597
dev_err(component->dev, "Failed to issue reset: %d\n", ret);
610
598
return ret;
611
599
}
600
+
601
+
regmap_register_patch(ssm2602->regmap, ssm2602_patch,
602
+
ARRAY_SIZE(ssm2602_patch));
612
603
613
604
/* set the update bits */
614
605
regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
+2
-2
sound/soc/dwc/dwc-i2s.c
+2
-2
sound/soc/dwc/dwc-i2s.c
···
133
133
134
134
/* Error Handling: TX */
135
135
if (isr[i] & ISR_TXFO) {
136
-
dev_err(dev->dev, "TX overrun (ch_id=%d)\n", i);
136
+
dev_err_ratelimited(dev->dev, "TX overrun (ch_id=%d)\n", i);
137
137
irq_valid = true;
138
138
}
139
139
140
140
/* Error Handling: TX */
141
141
if (isr[i] & ISR_RXFO) {
142
-
dev_err(dev->dev, "RX overrun (ch_id=%d)\n", i);
142
+
dev_err_ratelimited(dev->dev, "RX overrun (ch_id=%d)\n", i);
143
143
irq_valid = true;
144
144
}
145
145
}
+13
-1
sound/soc/fsl/fsl_micfil.c
+13
-1
sound/soc/fsl/fsl_micfil.c
···
1159
1159
ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
1160
1160
if (ret) {
1161
1161
dev_err(&pdev->dev, "failed to pcm register\n");
1162
-
return ret;
1162
+
goto err_pm_disable;
1163
1163
}
1164
1164
1165
1165
fsl_micfil_dai.capture.formats = micfil->soc->formats;
···
1169
1169
if (ret) {
1170
1170
dev_err(&pdev->dev, "failed to register component %s\n",
1171
1171
fsl_micfil_component.name);
1172
+
goto err_pm_disable;
1172
1173
}
1173
1174
1174
1175
return ret;
1176
+
1177
+
err_pm_disable:
1178
+
pm_runtime_disable(&pdev->dev);
1179
+
1180
+
return ret;
1181
+
}
1182
+
1183
+
static void fsl_micfil_remove(struct platform_device *pdev)
1184
+
{
1185
+
pm_runtime_disable(&pdev->dev);
1175
1186
}
1176
1187
1177
1188
static int __maybe_unused fsl_micfil_runtime_suspend(struct device *dev)
···
1243
1232
1244
1233
static struct platform_driver fsl_micfil_driver = {
1245
1234
.probe = fsl_micfil_probe,
1235
+
.remove_new = fsl_micfil_remove,
1246
1236
.driver = {
1247
1237
.name = "fsl-micfil-dai",
1248
1238
.pm = &fsl_micfil_pm_ops,
+50
-4
sound/soc/jz4740/jz4740-i2s.c
+50
-4
sound/soc/jz4740/jz4740-i2s.c
···
218
218
return 0;
219
219
}
220
220
221
+
static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate,
222
+
unsigned long i2sdiv_max)
223
+
{
224
+
unsigned long div, rate1, rate2, err1, err2;
225
+
226
+
div = mclk / (64 * rate);
227
+
if (div == 0)
228
+
div = 1;
229
+
230
+
rate1 = mclk / (64 * div);
231
+
rate2 = mclk / (64 * (div + 1));
232
+
233
+
err1 = abs(rate1 - rate);
234
+
err2 = abs(rate2 - rate);
235
+
236
+
/*
237
+
* Choose the divider that produces the smallest error in the
238
+
* output rate and reject dividers with a 5% or higher error.
239
+
* In the event that both dividers are outside the acceptable
240
+
* error margin, reject the rate to prevent distorted audio.
241
+
* (The number 5% is arbitrary.)
242
+
*/
243
+
if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20)
244
+
return div;
245
+
if (div < i2sdiv_max && err2 < rate/20)
246
+
return div + 1;
247
+
248
+
return -EINVAL;
249
+
}
250
+
221
251
static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
222
252
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
223
253
{
224
254
struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai);
225
255
struct regmap_field *div_field;
256
+
unsigned long i2sdiv_max;
226
257
unsigned int sample_size;
227
-
uint32_t ctrl;
228
-
int div;
258
+
uint32_t ctrl, conf;
259
+
int div = 1;
229
260
230
261
regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl);
231
-
232
-
div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params));
262
+
regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf);
233
263
234
264
switch (params_format(params)) {
235
265
case SNDRV_PCM_FORMAT_S8:
···
288
258
ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
289
259
290
260
div_field = i2s->field_i2sdiv_playback;
261
+
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb,
262
+
i2s->soc_info->field_i2sdiv_playback.lsb);
291
263
} else {
292
264
ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE;
293
265
ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size);
294
266
295
267
div_field = i2s->field_i2sdiv_capture;
268
+
i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb,
269
+
i2s->soc_info->field_i2sdiv_capture.lsb);
270
+
}
271
+
272
+
/*
273
+
* Only calculate I2SDIV if we're supplying the bit or frame clock.
274
+
* If the codec is supplying both clocks then the divider output is
275
+
* unused, and we don't want it to limit the allowed sample rates.
276
+
*/
277
+
if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) {
278
+
div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s),
279
+
params_rate(params), i2sdiv_max);
280
+
if (div < 0)
281
+
return div;
296
282
}
297
283
298
284
regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl);
+6
-1
sound/soc/sof/amd/acp-ipc.c
+6
-1
sound/soc/sof/amd/acp-ipc.c
···
209
209
acp_mailbox_read(sdev, offset, p, sz);
210
210
} else {
211
211
struct snd_pcm_substream *substream = sps->substream;
212
-
struct acp_dsp_stream *stream = substream->runtime->private_data;
212
+
struct acp_dsp_stream *stream;
213
+
214
+
if (!substream || !substream->runtime)
215
+
return -ESTRPIPE;
216
+
217
+
stream = substream->runtime->private_data;
213
218
214
219
if (!stream)
215
220
return -ESTRPIPE;