1 // SPDX-License-Identifier: GPL-2.0
3 // Freescale Generic ASoC Sound Card driver with ASRC
5 // Copyright (C) 2014 Freescale Semiconductor, Inc.
7 // Author: Nicolin Chen <nicoleotsuka@gmail.com>
10 #include <linux/i2c.h>
11 #include <linux/module.h>
12 #include <linux/of_platform.h>
13 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
14 #include <sound/ac97_codec.h>
16 #include <sound/pcm_params.h>
17 #include <sound/soc.h>
18 #include <sound/jack.h>
19 #include <sound/simple_card_utils.h>
23 #include "imx-audmux.h"
25 #include "../codecs/sgtl5000.h"
26 #include "../codecs/wm8962.h"
27 #include "../codecs/wm8960.h"
29 #define CS427x_SYSCLK_MCLK 0
34 /* Default DAI format without Master and Slave flag */
35 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
38 * struct codec_priv - CODEC private data
39 * @mclk_freq: Clock rate of MCLK
40 * @mclk_id: MCLK (or main clock) id for set_sysclk()
41 * @fll_id: FLL (or secordary clock) id for set_sysclk()
42 * @pll_id: PLL id for set_pll()
45 unsigned long mclk_freq;
52 * struct cpu_priv - CPU private data
53 * @sysclk_freq: SYSCLK rates for set_sysclk()
54 * @sysclk_dir: SYSCLK directions for set_sysclk()
55 * @sysclk_id: SYSCLK ids for set_sysclk()
56 * @slot_width: Slot width of each frame
58 * Note: [1] for tx and [0] for rx
61 unsigned long sysclk_freq[2];
68 * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
69 * @dai_link: DAI link structure including normal one and DPCM link
70 * @hp_jack: Headphone Jack structure
71 * @mic_jack: Microphone Jack structure
72 * @pdev: platform device pointer
73 * @codec_priv: CODEC private data
74 * @cpu_priv: CPU private data
75 * @card: ASoC card structure
76 * @streams: Mask of current active streams
77 * @sample_rate: Current sample rate
78 * @sample_format: Current sample format
79 * @asrc_rate: ASRC sample rate used by Back-Ends
80 * @asrc_format: ASRC sample format used by Back-Ends
81 * @dai_fmt: DAI format between CPU and CODEC
85 struct fsl_asoc_card_priv {
86 struct snd_soc_dai_link dai_link[3];
87 struct asoc_simple_jack hp_jack;
88 struct asoc_simple_jack mic_jack;
89 struct platform_device *pdev;
90 struct codec_priv codec_priv;
91 struct cpu_priv cpu_priv;
92 struct snd_soc_card card;
95 snd_pcm_format_t sample_format;
97 snd_pcm_format_t asrc_format;
103 * This dapm route map exists for DPCM link only.
104 * The other routes shall go through Device Tree.
106 * Note: keep all ASRC routes in the second half
107 * to drop them easily for non-ASRC cases.
109 static const struct snd_soc_dapm_route audio_map[] = {
110 /* 1st half -- Normal DAPM routes */
111 {"Playback", NULL, "CPU-Playback"},
112 {"CPU-Capture", NULL, "Capture"},
113 /* 2nd half -- ASRC DAPM routes */
114 {"CPU-Playback", NULL, "ASRC-Playback"},
115 {"ASRC-Capture", NULL, "CPU-Capture"},
118 static const struct snd_soc_dapm_route audio_map_ac97[] = {
119 /* 1st half -- Normal DAPM routes */
120 {"Playback", NULL, "AC97 Playback"},
121 {"AC97 Capture", NULL, "Capture"},
122 /* 2nd half -- ASRC DAPM routes */
123 {"AC97 Playback", NULL, "ASRC-Playback"},
124 {"ASRC-Capture", NULL, "AC97 Capture"},
127 static const struct snd_soc_dapm_route audio_map_tx[] = {
128 /* 1st half -- Normal DAPM routes */
129 {"Playback", NULL, "CPU-Playback"},
130 /* 2nd half -- ASRC DAPM routes */
131 {"CPU-Playback", NULL, "ASRC-Playback"},
134 /* Add all possible widgets into here without being redundant */
135 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
136 SND_SOC_DAPM_LINE("Line Out Jack", NULL),
137 SND_SOC_DAPM_LINE("Line In Jack", NULL),
138 SND_SOC_DAPM_HP("Headphone Jack", NULL),
139 SND_SOC_DAPM_SPK("Ext Spk", NULL),
140 SND_SOC_DAPM_MIC("Mic Jack", NULL),
141 SND_SOC_DAPM_MIC("AMIC", NULL),
142 SND_SOC_DAPM_MIC("DMIC", NULL),
145 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
147 return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
150 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
151 struct snd_pcm_hw_params *params)
153 struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
154 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
155 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
156 struct codec_priv *codec_priv = &priv->codec_priv;
157 struct cpu_priv *cpu_priv = &priv->cpu_priv;
158 struct device *dev = rtd->card->dev;
159 unsigned int pll_out;
162 priv->sample_rate = params_rate(params);
163 priv->sample_format = params_format(params);
164 priv->streams |= BIT(substream->stream);
166 if (fsl_asoc_card_is_ac97(priv))
169 /* Specific configurations of DAIs starts from here */
170 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
171 cpu_priv->sysclk_freq[tx],
172 cpu_priv->sysclk_dir[tx]);
173 if (ret && ret != -ENOTSUPP) {
174 dev_err(dev, "failed to set sysclk for cpu dai\n");
178 if (cpu_priv->slot_width) {
179 ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), 0x3, 0x3, 2,
180 cpu_priv->slot_width);
181 if (ret && ret != -ENOTSUPP) {
182 dev_err(dev, "failed to set TDM slot for cpu dai\n");
187 /* Specific configuration for PLL */
188 if (codec_priv->pll_id && codec_priv->fll_id) {
189 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
190 pll_out = priv->sample_rate * 384;
192 pll_out = priv->sample_rate * 256;
194 ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
197 codec_priv->mclk_freq, pll_out);
199 dev_err(dev, "failed to start FLL: %d\n", ret);
203 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
205 pll_out, SND_SOC_CLOCK_IN);
207 if (ret && ret != -ENOTSUPP) {
208 dev_err(dev, "failed to set SYSCLK: %d\n", ret);
216 priv->streams &= ~BIT(substream->stream);
220 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
222 struct snd_soc_pcm_runtime *rtd = substream->private_data;
223 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
224 struct codec_priv *codec_priv = &priv->codec_priv;
225 struct device *dev = rtd->card->dev;
228 priv->streams &= ~BIT(substream->stream);
230 if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) {
231 /* Force freq to be 0 to avoid error message in codec */
232 ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0),
237 dev_err(dev, "failed to switch away from FLL: %d\n", ret);
241 ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0),
242 codec_priv->pll_id, 0, 0, 0);
243 if (ret && ret != -ENOTSUPP) {
244 dev_err(dev, "failed to stop FLL: %d\n", ret);
252 static const struct snd_soc_ops fsl_asoc_card_ops = {
253 .hw_params = fsl_asoc_card_hw_params,
254 .hw_free = fsl_asoc_card_hw_free,
257 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
258 struct snd_pcm_hw_params *params)
260 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
261 struct snd_interval *rate;
262 struct snd_mask *mask;
264 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
265 rate->max = rate->min = priv->asrc_rate;
267 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
269 snd_mask_set_format(mask, priv->asrc_format);
274 SND_SOC_DAILINK_DEFS(hifi,
275 DAILINK_COMP_ARRAY(COMP_EMPTY()),
276 DAILINK_COMP_ARRAY(COMP_EMPTY()),
277 DAILINK_COMP_ARRAY(COMP_EMPTY()));
279 SND_SOC_DAILINK_DEFS(hifi_fe,
280 DAILINK_COMP_ARRAY(COMP_EMPTY()),
281 DAILINK_COMP_ARRAY(COMP_DUMMY()),
282 DAILINK_COMP_ARRAY(COMP_EMPTY()));
284 SND_SOC_DAILINK_DEFS(hifi_be,
285 DAILINK_COMP_ARRAY(COMP_EMPTY()),
286 DAILINK_COMP_ARRAY(COMP_EMPTY()),
287 DAILINK_COMP_ARRAY(COMP_DUMMY()));
289 static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
290 /* Default ASoC DAI Link*/
293 .stream_name = "HiFi",
294 .ops = &fsl_asoc_card_ops,
295 SND_SOC_DAILINK_REG(hifi),
297 /* DPCM Link between Front-End and Back-End (Optional) */
299 .name = "HiFi-ASRC-FE",
300 .stream_name = "HiFi-ASRC-FE",
304 SND_SOC_DAILINK_REG(hifi_fe),
307 .name = "HiFi-ASRC-BE",
308 .stream_name = "HiFi-ASRC-BE",
309 .be_hw_params_fixup = be_hw_params_fixup,
310 .ops = &fsl_asoc_card_ops,
314 SND_SOC_DAILINK_REG(hifi_be),
318 static int fsl_asoc_card_audmux_init(struct device_node *np,
319 struct fsl_asoc_card_priv *priv)
321 struct device *dev = &priv->pdev->dev;
322 u32 int_ptcr = 0, ext_ptcr = 0;
323 int int_port, ext_port;
326 ret = of_property_read_u32(np, "mux-int-port", &int_port);
328 dev_err(dev, "mux-int-port missing or invalid\n");
331 ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
333 dev_err(dev, "mux-ext-port missing or invalid\n");
338 * The port numbering in the hardware manual starts at 1, while
339 * the AUDMUX API expects it starts at 0.
345 * Use asynchronous mode (6 wires) for all cases except AC97.
346 * If only 4 wires are needed, just set SSI into
347 * synchronous mode and enable 4 PADs in IOMUX.
349 switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
350 case SND_SOC_DAIFMT_CBM_CFM:
351 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
352 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
353 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
354 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
355 IMX_AUDMUX_V2_PTCR_RFSDIR |
356 IMX_AUDMUX_V2_PTCR_RCLKDIR |
357 IMX_AUDMUX_V2_PTCR_TFSDIR |
358 IMX_AUDMUX_V2_PTCR_TCLKDIR;
360 case SND_SOC_DAIFMT_CBM_CFS:
361 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
362 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
363 IMX_AUDMUX_V2_PTCR_RCLKDIR |
364 IMX_AUDMUX_V2_PTCR_TCLKDIR;
365 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
366 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
367 IMX_AUDMUX_V2_PTCR_RFSDIR |
368 IMX_AUDMUX_V2_PTCR_TFSDIR;
370 case SND_SOC_DAIFMT_CBS_CFM:
371 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
372 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
373 IMX_AUDMUX_V2_PTCR_RFSDIR |
374 IMX_AUDMUX_V2_PTCR_TFSDIR;
375 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
376 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
377 IMX_AUDMUX_V2_PTCR_RCLKDIR |
378 IMX_AUDMUX_V2_PTCR_TCLKDIR;
380 case SND_SOC_DAIFMT_CBS_CFS:
381 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
382 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
383 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
384 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
385 IMX_AUDMUX_V2_PTCR_RFSDIR |
386 IMX_AUDMUX_V2_PTCR_RCLKDIR |
387 IMX_AUDMUX_V2_PTCR_TFSDIR |
388 IMX_AUDMUX_V2_PTCR_TCLKDIR;
391 if (!fsl_asoc_card_is_ac97(priv))
395 if (fsl_asoc_card_is_ac97(priv)) {
396 int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
397 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
398 IMX_AUDMUX_V2_PTCR_TCLKDIR;
399 ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
400 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
401 IMX_AUDMUX_V2_PTCR_TFSDIR;
404 /* Asynchronous mode can not be set along with RCLKDIR */
405 if (!fsl_asoc_card_is_ac97(priv)) {
407 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
409 ret = imx_audmux_v2_configure_port(int_port, 0,
412 dev_err(dev, "audmux internal port setup failed\n");
417 ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
418 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
420 dev_err(dev, "audmux internal port setup failed\n");
424 if (!fsl_asoc_card_is_ac97(priv)) {
426 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
428 ret = imx_audmux_v2_configure_port(ext_port, 0,
431 dev_err(dev, "audmux external port setup failed\n");
436 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
437 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
439 dev_err(dev, "audmux external port setup failed\n");
446 static int hp_jack_event(struct notifier_block *nb, unsigned long event,
449 struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
450 struct snd_soc_dapm_context *dapm = &jack->card->dapm;
452 if (event & SND_JACK_HEADPHONE)
453 /* Disable speaker if headphone is plugged in */
454 snd_soc_dapm_disable_pin(dapm, "Ext Spk");
456 snd_soc_dapm_enable_pin(dapm, "Ext Spk");
461 static struct notifier_block hp_jack_nb = {
462 .notifier_call = hp_jack_event,
465 static int mic_jack_event(struct notifier_block *nb, unsigned long event,
468 struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
469 struct snd_soc_dapm_context *dapm = &jack->card->dapm;
471 if (event & SND_JACK_MICROPHONE)
472 /* Disable dmic if microphone is plugged in */
473 snd_soc_dapm_disable_pin(dapm, "DMIC");
475 snd_soc_dapm_enable_pin(dapm, "DMIC");
480 static struct notifier_block mic_jack_nb = {
481 .notifier_call = mic_jack_event,
484 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
486 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
487 struct snd_soc_pcm_runtime *rtd = list_first_entry(
488 &card->rtd_list, struct snd_soc_pcm_runtime, list);
489 struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
490 struct codec_priv *codec_priv = &priv->codec_priv;
491 struct device *dev = card->dev;
494 if (fsl_asoc_card_is_ac97(priv)) {
495 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
496 struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
497 struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
500 * Use slots 3/4 for S/PDIF so SSI won't try to enable
501 * other slots and send some samples there
502 * due to SLOTREQ bits for S/PDIF received from codec
504 snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
505 AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
511 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
512 codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
513 if (ret && ret != -ENOTSUPP) {
514 dev_err(dev, "failed to set sysclk in %s\n", __func__);
521 static int fsl_asoc_card_probe(struct platform_device *pdev)
523 struct device_node *cpu_np, *codec_np, *asrc_np;
524 struct device_node *np = pdev->dev.of_node;
525 struct platform_device *asrc_pdev = NULL;
526 struct device_node *bitclkmaster = NULL;
527 struct device_node *framemaster = NULL;
528 struct platform_device *cpu_pdev;
529 struct fsl_asoc_card_priv *priv;
530 struct device *codec_dev = NULL;
531 const char *codec_dai_name;
532 const char *codec_dev_name;
537 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
541 cpu_np = of_parse_phandle(np, "audio-cpu", 0);
542 /* Give a chance to old DT binding */
544 cpu_np = of_parse_phandle(np, "ssi-controller", 0);
546 dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
551 cpu_pdev = of_find_device_by_node(cpu_np);
553 dev_err(&pdev->dev, "failed to find CPU DAI device\n");
558 codec_np = of_parse_phandle(np, "audio-codec", 0);
560 struct platform_device *codec_pdev;
561 struct i2c_client *codec_i2c;
563 codec_i2c = of_find_i2c_device_by_node(codec_np);
565 codec_dev = &codec_i2c->dev;
566 codec_dev_name = codec_i2c->name;
569 codec_pdev = of_find_device_by_node(codec_np);
571 codec_dev = &codec_pdev->dev;
572 codec_dev_name = codec_pdev->name;
577 asrc_np = of_parse_phandle(np, "audio-asrc", 0);
579 asrc_pdev = of_find_device_by_node(asrc_np);
581 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
583 struct clk *codec_clk = clk_get(codec_dev, NULL);
585 if (!IS_ERR(codec_clk)) {
586 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
591 /* Default sample rate and format, will be updated in hw_params() */
592 priv->sample_rate = 44100;
593 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
595 /* Assign a default DAI format, and allow each card to overwrite it */
596 priv->dai_fmt = DAI_FMT_BASE;
598 memcpy(priv->dai_link, fsl_asoc_card_dai,
599 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
601 priv->card.dapm_routes = audio_map;
602 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
603 /* Diversify the card configurations */
604 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
605 codec_dai_name = "cs42888";
606 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
607 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
608 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
609 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
610 priv->cpu_priv.slot_width = 32;
611 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
612 } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
613 codec_dai_name = "cs4271-hifi";
614 priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
615 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
616 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
617 codec_dai_name = "sgtl5000";
618 priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
619 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
620 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
621 codec_dai_name = "wm8962";
622 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
623 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
624 priv->codec_priv.pll_id = WM8962_FLL;
625 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
626 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
627 codec_dai_name = "wm8960-hifi";
628 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
629 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
630 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
631 } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
632 codec_dai_name = "ac97-hifi";
633 priv->dai_fmt = SND_SOC_DAIFMT_AC97;
634 priv->card.dapm_routes = audio_map_ac97;
635 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
636 } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
637 codec_dai_name = "fsl-mqs-dai";
638 priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
639 SND_SOC_DAIFMT_CBS_CFS |
640 SND_SOC_DAIFMT_NB_NF;
641 priv->dai_link[1].dpcm_capture = 0;
642 priv->dai_link[2].dpcm_capture = 0;
643 priv->card.dapm_routes = audio_map_tx;
644 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
645 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
646 codec_dai_name = "wm8524-hifi";
647 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
648 priv->dai_link[1].dpcm_capture = 0;
649 priv->dai_link[2].dpcm_capture = 0;
650 priv->cpu_priv.slot_width = 32;
651 priv->card.dapm_routes = audio_map_tx;
652 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
654 dev_err(&pdev->dev, "unknown Device Tree compatible\n");
659 /* Format info from DT is optional. */
660 daifmt = snd_soc_of_parse_daifmt(np, NULL,
661 &bitclkmaster, &framemaster);
662 daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;
663 if (bitclkmaster || framemaster) {
664 if (codec_np == bitclkmaster)
665 daifmt |= (codec_np == framemaster) ?
666 SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS;
668 daifmt |= (codec_np == framemaster) ?
669 SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS;
671 /* Override dai_fmt with value from DT */
672 priv->dai_fmt = daifmt;
675 /* Change direction according to format */
676 if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) {
677 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
678 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
681 of_node_put(bitclkmaster);
682 of_node_put(framemaster);
684 if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
685 dev_err(&pdev->dev, "failed to find codec device\n");
690 /* Common settings for corresponding Freescale CPU DAI driver */
691 if (of_node_name_eq(cpu_np, "ssi")) {
692 /* Only SSI needs to configure AUDMUX */
693 ret = fsl_asoc_card_audmux_init(np, priv);
695 dev_err(&pdev->dev, "failed to init audmux\n");
698 } else if (of_node_name_eq(cpu_np, "esai")) {
699 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
700 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
701 } else if (of_node_name_eq(cpu_np, "sai")) {
702 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
703 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
706 /* Initialize sound card */
708 priv->card.dev = &pdev->dev;
709 ret = snd_soc_of_parse_card_name(&priv->card, "model");
711 snprintf(priv->name, sizeof(priv->name), "%s-audio",
712 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
713 priv->card.name = priv->name;
715 priv->card.dai_link = priv->dai_link;
716 priv->card.late_probe = fsl_asoc_card_late_probe;
717 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
718 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
720 /* Drop the second half of DAPM routes -- ASRC */
722 priv->card.num_dapm_routes /= 2;
724 if (of_property_read_bool(np, "audio-routing")) {
725 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
727 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
732 /* Normal DAI Link */
733 priv->dai_link[0].cpus->of_node = cpu_np;
734 priv->dai_link[0].codecs->dai_name = codec_dai_name;
736 if (!fsl_asoc_card_is_ac97(priv))
737 priv->dai_link[0].codecs->of_node = codec_np;
741 ret = of_property_read_u32(cpu_np, "cell-index", &idx);
744 "cannot get CPU index property\n");
748 priv->dai_link[0].codecs->name =
749 devm_kasprintf(&pdev->dev, GFP_KERNEL,
752 if (!priv->dai_link[0].codecs->name) {
758 priv->dai_link[0].platforms->of_node = cpu_np;
759 priv->dai_link[0].dai_fmt = priv->dai_fmt;
760 priv->card.num_links = 1;
763 /* DPCM DAI Links only if ASRC exsits */
764 priv->dai_link[1].cpus->of_node = asrc_np;
765 priv->dai_link[1].platforms->of_node = asrc_np;
766 priv->dai_link[2].codecs->dai_name = codec_dai_name;
767 priv->dai_link[2].codecs->of_node = codec_np;
768 priv->dai_link[2].codecs->name =
769 priv->dai_link[0].codecs->name;
770 priv->dai_link[2].cpus->of_node = cpu_np;
771 priv->dai_link[2].dai_fmt = priv->dai_fmt;
772 priv->card.num_links = 3;
774 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
777 dev_err(&pdev->dev, "failed to get output rate\n");
782 ret = of_property_read_u32(asrc_np, "fsl,asrc-format",
785 /* Fallback to old binding; translate to asrc_format */
786 ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
790 "failed to decide output format\n");
795 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
797 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
801 /* Finish card registering */
802 platform_set_drvdata(pdev, priv);
803 snd_soc_card_set_drvdata(&priv->card, priv);
805 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
807 if (ret != -EPROBE_DEFER)
808 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
813 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
814 * asoc_simple_init_jack uses these properties for creating
815 * Headphone Jack and Microphone Jack.
817 * The notifier is initialized in snd_soc_card_jack_new(), then
818 * snd_soc_jack_notifier_register can be called.
820 if (of_property_read_bool(np, "hp-det-gpio")) {
821 ret = asoc_simple_init_jack(&priv->card, &priv->hp_jack,
822 1, NULL, "Headphone Jack");
826 snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
829 if (of_property_read_bool(np, "mic-det-gpio")) {
830 ret = asoc_simple_init_jack(&priv->card, &priv->mic_jack,
831 0, NULL, "Mic Jack");
835 snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
839 of_node_put(asrc_np);
840 of_node_put(codec_np);
841 put_device(&cpu_pdev->dev);
848 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
849 { .compatible = "fsl,imx-audio-ac97", },
850 { .compatible = "fsl,imx-audio-cs42888", },
851 { .compatible = "fsl,imx-audio-cs427x", },
852 { .compatible = "fsl,imx-audio-sgtl5000", },
853 { .compatible = "fsl,imx-audio-wm8962", },
854 { .compatible = "fsl,imx-audio-wm8960", },
855 { .compatible = "fsl,imx-audio-mqs", },
856 { .compatible = "fsl,imx-audio-wm8524", },
859 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
861 static struct platform_driver fsl_asoc_card_driver = {
862 .probe = fsl_asoc_card_probe,
864 .name = "fsl-asoc-card",
865 .pm = &snd_soc_pm_ops,
866 .of_match_table = fsl_asoc_card_dt_ids,
869 module_platform_driver(fsl_asoc_card_driver);
871 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
872 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
873 MODULE_ALIAS("platform:fsl-asoc-card");
874 MODULE_LICENSE("GPL");