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"
28 #include "../codecs/wm8994.h"
29 #include "../codecs/tlv320aic31xx.h"
30 #include "../codecs/nau8822.h"
32 #define DRIVER_NAME "fsl-asoc-card"
34 #define CS427x_SYSCLK_MCLK 0
39 /* Default DAI format without Master and Slave flag */
40 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
43 * struct codec_priv - CODEC private data
44 * @mclk: Main clock of the CODEC
45 * @mclk_freq: Clock rate of MCLK
46 * @free_freq: Clock rate of MCLK for hw_free()
47 * @mclk_id: MCLK (or main clock) id for set_sysclk()
48 * @fll_id: FLL (or secordary clock) id for set_sysclk()
49 * @pll_id: PLL id for set_pll()
53 unsigned long mclk_freq;
54 unsigned long free_freq;
61 * struct cpu_priv - CPU private data
62 * @sysclk_freq: SYSCLK rates for set_sysclk()
63 * @sysclk_dir: SYSCLK directions for set_sysclk()
64 * @sysclk_id: SYSCLK ids for set_sysclk()
65 * @slot_width: Slot width of each frame
66 * @slot_num: Number of slots of each frame
68 * Note: [1] for tx and [0] for rx
71 unsigned long sysclk_freq[2];
79 * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data
80 * @dai_link: DAI link structure including normal one and DPCM link
81 * @hp_jack: Headphone Jack structure
82 * @mic_jack: Microphone Jack structure
83 * @pdev: platform device pointer
84 * @codec_priv: CODEC private data
85 * @cpu_priv: CPU private data
86 * @card: ASoC card structure
87 * @streams: Mask of current active streams
88 * @sample_rate: Current sample rate
89 * @sample_format: Current sample format
90 * @asrc_rate: ASRC sample rate used by Back-Ends
91 * @asrc_format: ASRC sample format used by Back-Ends
92 * @dai_fmt: DAI format between CPU and CODEC
96 struct fsl_asoc_card_priv {
97 struct snd_soc_dai_link dai_link[3];
98 struct simple_util_jack hp_jack;
99 struct simple_util_jack mic_jack;
100 struct platform_device *pdev;
101 struct codec_priv codec_priv;
102 struct cpu_priv cpu_priv;
103 struct snd_soc_card card;
106 snd_pcm_format_t sample_format;
108 snd_pcm_format_t asrc_format;
114 * This dapm route map exists for DPCM link only.
115 * The other routes shall go through Device Tree.
117 * Note: keep all ASRC routes in the second half
118 * to drop them easily for non-ASRC cases.
120 static const struct snd_soc_dapm_route audio_map[] = {
121 /* 1st half -- Normal DAPM routes */
122 {"Playback", NULL, "CPU-Playback"},
123 {"CPU-Capture", NULL, "Capture"},
124 /* 2nd half -- ASRC DAPM routes */
125 {"CPU-Playback", NULL, "ASRC-Playback"},
126 {"ASRC-Capture", NULL, "CPU-Capture"},
129 static const struct snd_soc_dapm_route audio_map_ac97[] = {
130 /* 1st half -- Normal DAPM routes */
131 {"AC97 Playback", NULL, "CPU AC97 Playback"},
132 {"CPU AC97 Capture", NULL, "AC97 Capture"},
133 /* 2nd half -- ASRC DAPM routes */
134 {"CPU AC97 Playback", NULL, "ASRC-Playback"},
135 {"ASRC-Capture", NULL, "CPU AC97 Capture"},
138 static const struct snd_soc_dapm_route audio_map_tx[] = {
139 /* 1st half -- Normal DAPM routes */
140 {"Playback", NULL, "CPU-Playback"},
141 /* 2nd half -- ASRC DAPM routes */
142 {"CPU-Playback", NULL, "ASRC-Playback"},
145 static const struct snd_soc_dapm_route audio_map_rx[] = {
146 /* 1st half -- Normal DAPM routes */
147 {"CPU-Capture", NULL, "Capture"},
148 /* 2nd half -- ASRC DAPM routes */
149 {"ASRC-Capture", NULL, "CPU-Capture"},
152 /* Add all possible widgets into here without being redundant */
153 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
154 SND_SOC_DAPM_LINE("Line Out Jack", NULL),
155 SND_SOC_DAPM_LINE("Line In Jack", NULL),
156 SND_SOC_DAPM_HP("Headphone Jack", NULL),
157 SND_SOC_DAPM_SPK("Ext Spk", NULL),
158 SND_SOC_DAPM_MIC("Mic Jack", NULL),
159 SND_SOC_DAPM_MIC("AMIC", NULL),
160 SND_SOC_DAPM_MIC("DMIC", NULL),
163 static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
165 return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
168 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
169 struct snd_pcm_hw_params *params)
171 struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
172 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
173 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
174 struct codec_priv *codec_priv = &priv->codec_priv;
175 struct cpu_priv *cpu_priv = &priv->cpu_priv;
176 struct device *dev = rtd->card->dev;
177 unsigned int pll_out;
180 priv->sample_rate = params_rate(params);
181 priv->sample_format = params_format(params);
182 priv->streams |= BIT(substream->stream);
184 if (fsl_asoc_card_is_ac97(priv))
187 /* Specific configurations of DAIs starts from here */
188 ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), cpu_priv->sysclk_id[tx],
189 cpu_priv->sysclk_freq[tx],
190 cpu_priv->sysclk_dir[tx]);
191 if (ret && ret != -ENOTSUPP) {
192 dev_err(dev, "failed to set sysclk for cpu dai\n");
196 if (cpu_priv->slot_width) {
197 if (!cpu_priv->slot_num)
198 cpu_priv->slot_num = 2;
200 ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_cpu(rtd, 0), 0x3, 0x3,
202 cpu_priv->slot_width);
203 if (ret && ret != -ENOTSUPP) {
204 dev_err(dev, "failed to set TDM slot for cpu dai\n");
209 /* Specific configuration for PLL */
210 if (codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
211 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
212 pll_out = priv->sample_rate * 384;
214 pll_out = priv->sample_rate * 256;
216 ret = snd_soc_dai_set_pll(snd_soc_rtd_to_codec(rtd, 0),
219 codec_priv->mclk_freq, pll_out);
221 dev_err(dev, "failed to start FLL: %d\n", ret);
225 ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0),
227 pll_out, SND_SOC_CLOCK_IN);
229 if (ret && ret != -ENOTSUPP) {
230 dev_err(dev, "failed to set SYSCLK: %d\n", ret);
238 priv->streams &= ~BIT(substream->stream);
242 static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream)
244 struct snd_soc_pcm_runtime *rtd = substream->private_data;
245 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
246 struct codec_priv *codec_priv = &priv->codec_priv;
247 struct device *dev = rtd->card->dev;
250 priv->streams &= ~BIT(substream->stream);
252 if (!priv->streams && codec_priv->pll_id >= 0 && codec_priv->fll_id >= 0) {
253 /* Force freq to be free_freq to avoid error message in codec */
254 ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0),
256 codec_priv->free_freq,
259 dev_err(dev, "failed to switch away from FLL: %d\n", ret);
263 ret = snd_soc_dai_set_pll(snd_soc_rtd_to_codec(rtd, 0),
264 codec_priv->pll_id, 0, 0, 0);
265 if (ret && ret != -ENOTSUPP) {
266 dev_err(dev, "failed to stop FLL: %d\n", ret);
274 static const struct snd_soc_ops fsl_asoc_card_ops = {
275 .hw_params = fsl_asoc_card_hw_params,
276 .hw_free = fsl_asoc_card_hw_free,
279 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
280 struct snd_pcm_hw_params *params)
282 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
283 struct snd_interval *rate;
284 struct snd_mask *mask;
286 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
287 rate->max = rate->min = priv->asrc_rate;
289 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
291 snd_mask_set_format(mask, priv->asrc_format);
296 SND_SOC_DAILINK_DEFS(hifi,
297 DAILINK_COMP_ARRAY(COMP_EMPTY()),
298 DAILINK_COMP_ARRAY(COMP_EMPTY()),
299 DAILINK_COMP_ARRAY(COMP_EMPTY()));
301 SND_SOC_DAILINK_DEFS(hifi_fe,
302 DAILINK_COMP_ARRAY(COMP_EMPTY()),
303 DAILINK_COMP_ARRAY(COMP_DUMMY()),
304 DAILINK_COMP_ARRAY(COMP_EMPTY()));
306 SND_SOC_DAILINK_DEFS(hifi_be,
307 DAILINK_COMP_ARRAY(COMP_EMPTY()),
308 DAILINK_COMP_ARRAY(COMP_EMPTY()),
309 DAILINK_COMP_ARRAY(COMP_DUMMY()));
311 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
312 /* Default ASoC DAI Link*/
315 .stream_name = "HiFi",
316 .ops = &fsl_asoc_card_ops,
317 SND_SOC_DAILINK_REG(hifi),
319 /* DPCM Link between Front-End and Back-End (Optional) */
321 .name = "HiFi-ASRC-FE",
322 .stream_name = "HiFi-ASRC-FE",
326 SND_SOC_DAILINK_REG(hifi_fe),
329 .name = "HiFi-ASRC-BE",
330 .stream_name = "HiFi-ASRC-BE",
331 .be_hw_params_fixup = be_hw_params_fixup,
332 .ops = &fsl_asoc_card_ops,
336 SND_SOC_DAILINK_REG(hifi_be),
340 static int fsl_asoc_card_audmux_init(struct device_node *np,
341 struct fsl_asoc_card_priv *priv)
343 struct device *dev = &priv->pdev->dev;
344 u32 int_ptcr = 0, ext_ptcr = 0;
345 int int_port, ext_port;
348 ret = of_property_read_u32(np, "mux-int-port", &int_port);
350 dev_err(dev, "mux-int-port missing or invalid\n");
353 ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
355 dev_err(dev, "mux-ext-port missing or invalid\n");
360 * The port numbering in the hardware manual starts at 1, while
361 * the AUDMUX API expects it starts at 0.
367 * Use asynchronous mode (6 wires) for all cases except AC97.
368 * If only 4 wires are needed, just set SSI into
369 * synchronous mode and enable 4 PADs in IOMUX.
371 switch (priv->dai_fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
372 case SND_SOC_DAIFMT_CBP_CFP:
373 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
374 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
375 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
376 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
377 IMX_AUDMUX_V2_PTCR_RFSDIR |
378 IMX_AUDMUX_V2_PTCR_RCLKDIR |
379 IMX_AUDMUX_V2_PTCR_TFSDIR |
380 IMX_AUDMUX_V2_PTCR_TCLKDIR;
382 case SND_SOC_DAIFMT_CBP_CFC:
383 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
384 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
385 IMX_AUDMUX_V2_PTCR_RCLKDIR |
386 IMX_AUDMUX_V2_PTCR_TCLKDIR;
387 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
388 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
389 IMX_AUDMUX_V2_PTCR_RFSDIR |
390 IMX_AUDMUX_V2_PTCR_TFSDIR;
392 case SND_SOC_DAIFMT_CBC_CFP:
393 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
394 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
395 IMX_AUDMUX_V2_PTCR_RFSDIR |
396 IMX_AUDMUX_V2_PTCR_TFSDIR;
397 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
398 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
399 IMX_AUDMUX_V2_PTCR_RCLKDIR |
400 IMX_AUDMUX_V2_PTCR_TCLKDIR;
402 case SND_SOC_DAIFMT_CBC_CFC:
403 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
404 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
405 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
406 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
407 IMX_AUDMUX_V2_PTCR_RFSDIR |
408 IMX_AUDMUX_V2_PTCR_RCLKDIR |
409 IMX_AUDMUX_V2_PTCR_TFSDIR |
410 IMX_AUDMUX_V2_PTCR_TCLKDIR;
413 if (!fsl_asoc_card_is_ac97(priv))
417 if (fsl_asoc_card_is_ac97(priv)) {
418 int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
419 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
420 IMX_AUDMUX_V2_PTCR_TCLKDIR;
421 ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
422 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
423 IMX_AUDMUX_V2_PTCR_TFSDIR;
426 /* Asynchronous mode can not be set along with RCLKDIR */
427 if (!fsl_asoc_card_is_ac97(priv)) {
429 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
431 ret = imx_audmux_v2_configure_port(int_port, 0,
434 dev_err(dev, "audmux internal port setup failed\n");
439 ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
440 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
442 dev_err(dev, "audmux internal port setup failed\n");
446 if (!fsl_asoc_card_is_ac97(priv)) {
448 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
450 ret = imx_audmux_v2_configure_port(ext_port, 0,
453 dev_err(dev, "audmux external port setup failed\n");
458 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
459 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
461 dev_err(dev, "audmux external port setup failed\n");
468 static int hp_jack_event(struct notifier_block *nb, unsigned long event,
471 struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
472 struct snd_soc_dapm_context *dapm = &jack->card->dapm;
474 if (event & SND_JACK_HEADPHONE)
475 /* Disable speaker if headphone is plugged in */
476 return snd_soc_dapm_disable_pin(dapm, "Ext Spk");
478 return snd_soc_dapm_enable_pin(dapm, "Ext Spk");
481 static struct notifier_block hp_jack_nb = {
482 .notifier_call = hp_jack_event,
485 static int mic_jack_event(struct notifier_block *nb, unsigned long event,
488 struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
489 struct snd_soc_dapm_context *dapm = &jack->card->dapm;
491 if (event & SND_JACK_MICROPHONE)
492 /* Disable dmic if microphone is plugged in */
493 return snd_soc_dapm_disable_pin(dapm, "DMIC");
495 return snd_soc_dapm_enable_pin(dapm, "DMIC");
498 static struct notifier_block mic_jack_nb = {
499 .notifier_call = mic_jack_event,
502 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
504 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
505 struct snd_soc_pcm_runtime *rtd = list_first_entry(
506 &card->rtd_list, struct snd_soc_pcm_runtime, list);
507 struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
508 struct codec_priv *codec_priv = &priv->codec_priv;
509 struct device *dev = card->dev;
512 if (fsl_asoc_card_is_ac97(priv)) {
513 #if IS_ENABLED(CONFIG_SND_AC97_CODEC)
514 struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component;
515 struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
518 * Use slots 3/4 for S/PDIF so SSI won't try to enable
519 * other slots and send some samples there
520 * due to SLOTREQ bits for S/PDIF received from codec
522 snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
523 AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
529 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
530 codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
531 if (ret && ret != -ENOTSUPP) {
532 dev_err(dev, "failed to set sysclk in %s\n", __func__);
536 if (!IS_ERR_OR_NULL(codec_priv->mclk))
537 clk_prepare_enable(codec_priv->mclk);
542 static int fsl_asoc_card_probe(struct platform_device *pdev)
544 struct device_node *cpu_np, *codec_np, *asrc_np;
545 struct device_node *np = pdev->dev.of_node;
546 struct platform_device *asrc_pdev = NULL;
547 struct device_node *bitclkprovider = NULL;
548 struct device_node *frameprovider = NULL;
549 struct platform_device *cpu_pdev;
550 struct fsl_asoc_card_priv *priv;
551 struct device *codec_dev = NULL;
552 const char *codec_dai_name;
553 const char *codec_dev_name;
558 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
562 cpu_np = of_parse_phandle(np, "audio-cpu", 0);
563 /* Give a chance to old DT binding */
565 cpu_np = of_parse_phandle(np, "ssi-controller", 0);
567 dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
572 cpu_pdev = of_find_device_by_node(cpu_np);
574 dev_err(&pdev->dev, "failed to find CPU DAI device\n");
579 codec_np = of_parse_phandle(np, "audio-codec", 0);
581 struct platform_device *codec_pdev;
582 struct i2c_client *codec_i2c;
584 codec_i2c = of_find_i2c_device_by_node(codec_np);
586 codec_dev = &codec_i2c->dev;
587 codec_dev_name = codec_i2c->name;
590 codec_pdev = of_find_device_by_node(codec_np);
592 codec_dev = &codec_pdev->dev;
593 codec_dev_name = codec_pdev->name;
598 asrc_np = of_parse_phandle(np, "audio-asrc", 0);
600 asrc_pdev = of_find_device_by_node(asrc_np);
602 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
604 struct clk *codec_clk = clk_get(codec_dev, NULL);
606 if (!IS_ERR(codec_clk)) {
607 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
612 /* Default sample rate and format, will be updated in hw_params() */
613 priv->sample_rate = 44100;
614 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
616 /* Assign a default DAI format, and allow each card to overwrite it */
617 priv->dai_fmt = DAI_FMT_BASE;
619 memcpy(priv->dai_link, fsl_asoc_card_dai,
620 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
622 priv->card.dapm_routes = audio_map;
623 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
624 priv->card.driver_name = DRIVER_NAME;
626 priv->codec_priv.fll_id = -1;
627 priv->codec_priv.pll_id = -1;
629 /* Diversify the card configurations */
630 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
631 codec_dai_name = "cs42888";
632 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
633 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
634 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
635 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
636 priv->cpu_priv.slot_width = 32;
637 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
638 } else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
639 codec_dai_name = "cs4271-hifi";
640 priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
641 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
642 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
643 codec_dai_name = "sgtl5000";
644 priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
645 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
646 } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) {
647 codec_dai_name = "tlv320aic32x4-hifi";
648 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
649 } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
650 codec_dai_name = "tlv320dac31xx-hifi";
651 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
652 priv->dai_link[1].dpcm_capture = 0;
653 priv->dai_link[2].dpcm_capture = 0;
654 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
655 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
656 priv->card.dapm_routes = audio_map_tx;
657 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
658 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
659 codec_dai_name = "wm8962";
660 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
661 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
662 priv->codec_priv.pll_id = WM8962_FLL;
663 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
664 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
665 codec_dai_name = "wm8960-hifi";
666 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
667 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
668 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
669 } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
670 codec_dai_name = "ac97-hifi";
671 priv->dai_fmt = SND_SOC_DAIFMT_AC97;
672 priv->card.dapm_routes = audio_map_ac97;
673 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
674 } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
675 codec_dai_name = "fsl-mqs-dai";
676 priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
677 SND_SOC_DAIFMT_CBC_CFC |
678 SND_SOC_DAIFMT_NB_NF;
679 priv->dai_link[1].dpcm_capture = 0;
680 priv->dai_link[2].dpcm_capture = 0;
681 priv->card.dapm_routes = audio_map_tx;
682 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
683 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
684 codec_dai_name = "wm8524-hifi";
685 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
686 priv->dai_link[1].dpcm_capture = 0;
687 priv->dai_link[2].dpcm_capture = 0;
688 priv->cpu_priv.slot_width = 32;
689 priv->card.dapm_routes = audio_map_tx;
690 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
691 } else if (of_device_is_compatible(np, "fsl,imx-audio-si476x")) {
692 codec_dai_name = "si476x-codec";
693 priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
694 priv->card.dapm_routes = audio_map_rx;
695 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_rx);
696 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8958")) {
697 codec_dai_name = "wm8994-aif1";
698 priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP;
699 priv->codec_priv.mclk_id = WM8994_FLL_SRC_MCLK1;
700 priv->codec_priv.fll_id = WM8994_SYSCLK_FLL1;
701 priv->codec_priv.pll_id = WM8994_FLL1;
702 priv->codec_priv.free_freq = priv->codec_priv.mclk_freq;
703 priv->card.dapm_routes = NULL;
704 priv->card.num_dapm_routes = 0;
705 } else if (of_device_is_compatible(np, "fsl,imx-audio-nau8822")) {
706 codec_dai_name = "nau8822-hifi";
707 priv->codec_priv.mclk_id = NAU8822_CLK_MCLK;
708 priv->codec_priv.fll_id = NAU8822_CLK_PLL;
709 priv->codec_priv.pll_id = NAU8822_CLK_PLL;
710 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
712 priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL);
714 dev_err(&pdev->dev, "unknown Device Tree compatible\n");
720 * Allow setting mclk-id from the device-tree node. Otherwise, the
721 * default value for each card configuration is used.
723 of_property_read_u32(np, "mclk-id", &priv->codec_priv.mclk_id);
725 /* Format info from DT is optional. */
726 snd_soc_daifmt_parse_clock_provider_as_phandle(np, NULL, &bitclkprovider, &frameprovider);
727 if (bitclkprovider || frameprovider) {
728 unsigned int daifmt = snd_soc_daifmt_parse_format(np, NULL);
730 if (codec_np == bitclkprovider)
731 daifmt |= (codec_np == frameprovider) ?
732 SND_SOC_DAIFMT_CBP_CFP : SND_SOC_DAIFMT_CBP_CFC;
734 daifmt |= (codec_np == frameprovider) ?
735 SND_SOC_DAIFMT_CBC_CFP : SND_SOC_DAIFMT_CBC_CFC;
737 /* Override dai_fmt with value from DT */
738 priv->dai_fmt = daifmt;
741 /* Change direction according to format */
742 if (priv->dai_fmt & SND_SOC_DAIFMT_CBP_CFP) {
743 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN;
744 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN;
747 of_node_put(bitclkprovider);
748 of_node_put(frameprovider);
750 if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
751 dev_dbg(&pdev->dev, "failed to find codec device\n");
756 /* Common settings for corresponding Freescale CPU DAI driver */
757 if (of_node_name_eq(cpu_np, "ssi")) {
758 /* Only SSI needs to configure AUDMUX */
759 ret = fsl_asoc_card_audmux_init(np, priv);
761 dev_err(&pdev->dev, "failed to init audmux\n");
764 } else if (of_node_name_eq(cpu_np, "esai")) {
765 struct clk *esai_clk = clk_get(&cpu_pdev->dev, "extal");
767 if (!IS_ERR(esai_clk)) {
768 priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(esai_clk);
769 priv->cpu_priv.sysclk_freq[RX] = clk_get_rate(esai_clk);
771 } else if (PTR_ERR(esai_clk) == -EPROBE_DEFER) {
776 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
777 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
778 } else if (of_node_name_eq(cpu_np, "sai")) {
779 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
780 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
783 /* Initialize sound card */
785 priv->card.dev = &pdev->dev;
786 priv->card.owner = THIS_MODULE;
787 ret = snd_soc_of_parse_card_name(&priv->card, "model");
789 snprintf(priv->name, sizeof(priv->name), "%s-audio",
790 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
791 priv->card.name = priv->name;
793 priv->card.dai_link = priv->dai_link;
794 priv->card.late_probe = fsl_asoc_card_late_probe;
795 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
796 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
798 /* Drop the second half of DAPM routes -- ASRC */
800 priv->card.num_dapm_routes /= 2;
802 if (of_property_read_bool(np, "audio-routing")) {
803 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
805 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
810 /* Normal DAI Link */
811 priv->dai_link[0].cpus->of_node = cpu_np;
812 priv->dai_link[0].codecs->dai_name = codec_dai_name;
814 if (!fsl_asoc_card_is_ac97(priv))
815 priv->dai_link[0].codecs->of_node = codec_np;
819 ret = of_property_read_u32(cpu_np, "cell-index", &idx);
822 "cannot get CPU index property\n");
826 priv->dai_link[0].codecs->name =
827 devm_kasprintf(&pdev->dev, GFP_KERNEL,
830 if (!priv->dai_link[0].codecs->name) {
836 priv->dai_link[0].platforms->of_node = cpu_np;
837 priv->dai_link[0].dai_fmt = priv->dai_fmt;
838 priv->card.num_links = 1;
841 /* DPCM DAI Links only if ASRC exists */
842 priv->dai_link[1].cpus->of_node = asrc_np;
843 priv->dai_link[1].platforms->of_node = asrc_np;
844 priv->dai_link[2].codecs->dai_name = codec_dai_name;
845 priv->dai_link[2].codecs->of_node = codec_np;
846 priv->dai_link[2].codecs->name =
847 priv->dai_link[0].codecs->name;
848 priv->dai_link[2].cpus->of_node = cpu_np;
849 priv->dai_link[2].dai_fmt = priv->dai_fmt;
850 priv->card.num_links = 3;
852 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
855 dev_err(&pdev->dev, "failed to get output rate\n");
860 ret = of_property_read_u32(asrc_np, "fsl,asrc-format", &asrc_fmt);
861 priv->asrc_format = (__force snd_pcm_format_t)asrc_fmt;
863 /* Fallback to old binding; translate to asrc_format */
864 ret = of_property_read_u32(asrc_np, "fsl,asrc-width",
868 "failed to decide output format\n");
873 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
875 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
879 /* Finish card registering */
880 platform_set_drvdata(pdev, priv);
881 snd_soc_card_set_drvdata(&priv->card, priv);
883 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
885 dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n");
890 * Properties "hp-det-gpio" and "mic-det-gpio" are optional, and
891 * simple_util_init_jack() uses these properties for creating
892 * Headphone Jack and Microphone Jack.
894 * The notifier is initialized in snd_soc_card_jack_new(), then
895 * snd_soc_jack_notifier_register can be called.
897 if (of_property_read_bool(np, "hp-det-gpio")) {
898 ret = simple_util_init_jack(&priv->card, &priv->hp_jack,
899 1, NULL, "Headphone Jack");
903 snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
906 if (of_property_read_bool(np, "mic-det-gpio")) {
907 ret = simple_util_init_jack(&priv->card, &priv->mic_jack,
908 0, NULL, "Mic Jack");
912 snd_soc_jack_notifier_register(&priv->mic_jack.jack, &mic_jack_nb);
916 of_node_put(asrc_np);
917 of_node_put(codec_np);
918 put_device(&cpu_pdev->dev);
925 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
926 { .compatible = "fsl,imx-audio-ac97", },
927 { .compatible = "fsl,imx-audio-cs42888", },
928 { .compatible = "fsl,imx-audio-cs427x", },
929 { .compatible = "fsl,imx-audio-tlv320aic32x4", },
930 { .compatible = "fsl,imx-audio-tlv320aic31xx", },
931 { .compatible = "fsl,imx-audio-sgtl5000", },
932 { .compatible = "fsl,imx-audio-wm8962", },
933 { .compatible = "fsl,imx-audio-wm8960", },
934 { .compatible = "fsl,imx-audio-mqs", },
935 { .compatible = "fsl,imx-audio-wm8524", },
936 { .compatible = "fsl,imx-audio-si476x", },
937 { .compatible = "fsl,imx-audio-wm8958", },
938 { .compatible = "fsl,imx-audio-nau8822", },
941 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
943 static struct platform_driver fsl_asoc_card_driver = {
944 .probe = fsl_asoc_card_probe,
947 .pm = &snd_soc_pm_ops,
948 .of_match_table = fsl_asoc_card_dt_ids,
951 module_platform_driver(fsl_asoc_card_driver);
953 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
954 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
955 MODULE_ALIAS("platform:" DRIVER_NAME);
956 MODULE_LICENSE("GPL");