2 * Freescale Generic ASoC Sound Card driver with ASRC
4 * Copyright (C) 2014 Freescale Semiconductor, Inc.
6 * Author: Nicolin Chen <nicoleotsuka@gmail.com>
8 * This file is licensed under the terms of the GNU General Public License
9 * version 2. This program is licensed "as is" without any warranty of any
10 * kind, whether express or implied.
13 #include <linux/clk.h>
14 #include <linux/i2c.h>
15 #include <linux/module.h>
16 #include <linux/of_platform.h>
17 #include <sound/pcm_params.h>
18 #include <sound/soc.h>
22 #include "imx-audmux.h"
24 #include "../codecs/sgtl5000.h"
25 #include "../codecs/wm8962.h"
26 #include "../codecs/wm8960.h"
31 /* Default DAI format without Master and Slave flag */
32 #define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
37 * @mclk_freq: Clock rate of MCLK
38 * @mclk_id: MCLK (or main clock) id for set_sysclk()
39 * @fll_id: FLL (or secordary clock) id for set_sysclk()
40 * @pll_id: PLL id for set_pll()
43 unsigned long mclk_freq;
52 * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
53 * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
54 * @sysclk_id[2]: SYSCLK ids for set_sysclk()
55 * @slot_width: Slot width of each frame
57 * Note: [1] for tx and [0] for rx
60 unsigned long sysclk_freq[2];
67 * Freescale Generic ASOC card private data
69 * @dai_link[3]: DAI link structure including normal one and DPCM link
70 * @pdev: platform device pointer
71 * @codec_priv: CODEC private data
72 * @cpu_priv: CPU private data
73 * @card: ASoC card structure
74 * @sample_rate: Current sample rate
75 * @sample_format: Current sample format
76 * @asrc_rate: ASRC sample rate used by Back-Ends
77 * @asrc_format: ASRC sample format used by Back-Ends
78 * @dai_fmt: DAI format between CPU and CODEC
82 struct fsl_asoc_card_priv {
83 struct snd_soc_dai_link dai_link[3];
84 struct platform_device *pdev;
85 struct codec_priv codec_priv;
86 struct cpu_priv cpu_priv;
87 struct snd_soc_card card;
97 * This dapm route map exsits for DPCM link only.
98 * The other routes shall go through Device Tree.
100 static const struct snd_soc_dapm_route audio_map[] = {
101 {"CPU-Playback", NULL, "ASRC-Playback"},
102 {"Playback", NULL, "CPU-Playback"},
103 {"ASRC-Capture", NULL, "CPU-Capture"},
104 {"CPU-Capture", NULL, "Capture"},
107 /* Add all possible widgets into here without being redundant */
108 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
109 SND_SOC_DAPM_LINE("Line Out Jack", NULL),
110 SND_SOC_DAPM_LINE("Line In Jack", NULL),
111 SND_SOC_DAPM_HP("Headphone Jack", NULL),
112 SND_SOC_DAPM_SPK("Ext Spk", NULL),
113 SND_SOC_DAPM_MIC("Mic Jack", NULL),
114 SND_SOC_DAPM_MIC("AMIC", NULL),
115 SND_SOC_DAPM_MIC("DMIC", NULL),
118 static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
119 struct snd_pcm_hw_params *params)
121 struct snd_soc_pcm_runtime *rtd = substream->private_data;
122 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
123 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
124 struct cpu_priv *cpu_priv = &priv->cpu_priv;
125 struct device *dev = rtd->card->dev;
128 priv->sample_rate = params_rate(params);
129 priv->sample_format = params_format(params);
132 * If codec-dai is DAI Master and all configurations are already in the
133 * set_bias_level(), bypass the remaining settings in hw_params().
134 * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
136 if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM)
139 /* Specific configurations of DAIs starts from here */
140 ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
141 cpu_priv->sysclk_freq[tx],
142 cpu_priv->sysclk_dir[tx]);
144 dev_err(dev, "failed to set sysclk for cpu dai\n");
148 if (cpu_priv->slot_width) {
149 ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
150 cpu_priv->slot_width);
152 dev_err(dev, "failed to set TDM slot for cpu dai\n");
160 static struct snd_soc_ops fsl_asoc_card_ops = {
161 .hw_params = fsl_asoc_card_hw_params,
164 static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
165 struct snd_pcm_hw_params *params)
167 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
168 struct snd_interval *rate;
169 struct snd_mask *mask;
171 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
172 rate->max = rate->min = priv->asrc_rate;
174 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
176 snd_mask_set(mask, priv->asrc_format);
181 static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
182 /* Default ASoC DAI Link*/
185 .stream_name = "HiFi",
186 .ops = &fsl_asoc_card_ops,
188 /* DPCM Link between Front-End and Back-End (Optional) */
190 .name = "HiFi-ASRC-FE",
191 .stream_name = "HiFi-ASRC-FE",
192 .codec_name = "snd-soc-dummy",
193 .codec_dai_name = "snd-soc-dummy-dai",
199 .name = "HiFi-ASRC-BE",
200 .stream_name = "HiFi-ASRC-BE",
201 .platform_name = "snd-soc-dummy",
202 .be_hw_params_fixup = be_hw_params_fixup,
203 .ops = &fsl_asoc_card_ops,
210 static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
211 struct snd_soc_dapm_context *dapm,
212 enum snd_soc_bias_level level)
214 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
215 struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
216 struct codec_priv *codec_priv = &priv->codec_priv;
217 struct device *dev = card->dev;
218 unsigned int pll_out;
221 if (dapm->dev != codec_dai->dev)
225 case SND_SOC_BIAS_PREPARE:
226 if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
229 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
230 pll_out = priv->sample_rate * 384;
232 pll_out = priv->sample_rate * 256;
234 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
236 codec_priv->mclk_freq, pll_out);
238 dev_err(dev, "failed to start FLL: %d\n", ret);
242 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
243 pll_out, SND_SOC_CLOCK_IN);
245 dev_err(dev, "failed to set SYSCLK: %d\n", ret);
250 case SND_SOC_BIAS_STANDBY:
251 if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
254 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
255 codec_priv->mclk_freq,
258 dev_err(dev, "failed to switch away from FLL: %d\n", ret);
262 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
264 dev_err(dev, "failed to stop FLL: %d\n", ret);
276 static int fsl_asoc_card_audmux_init(struct device_node *np,
277 struct fsl_asoc_card_priv *priv)
279 struct device *dev = &priv->pdev->dev;
280 u32 int_ptcr = 0, ext_ptcr = 0;
281 int int_port, ext_port;
284 ret = of_property_read_u32(np, "mux-int-port", &int_port);
286 dev_err(dev, "mux-int-port missing or invalid\n");
289 ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
291 dev_err(dev, "mux-ext-port missing or invalid\n");
296 * The port numbering in the hardware manual starts at 1, while
297 * the AUDMUX API expects it starts at 0.
303 * Use asynchronous mode (6 wires) for all cases.
304 * If only 4 wires are needed, just set SSI into
305 * synchronous mode and enable 4 PADs in IOMUX.
307 switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
308 case SND_SOC_DAIFMT_CBM_CFM:
309 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
310 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
311 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
312 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
313 IMX_AUDMUX_V2_PTCR_RFSDIR |
314 IMX_AUDMUX_V2_PTCR_RCLKDIR |
315 IMX_AUDMUX_V2_PTCR_TFSDIR |
316 IMX_AUDMUX_V2_PTCR_TCLKDIR;
318 case SND_SOC_DAIFMT_CBM_CFS:
319 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
320 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
321 IMX_AUDMUX_V2_PTCR_RCLKDIR |
322 IMX_AUDMUX_V2_PTCR_TCLKDIR;
323 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
324 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
325 IMX_AUDMUX_V2_PTCR_RFSDIR |
326 IMX_AUDMUX_V2_PTCR_TFSDIR;
328 case SND_SOC_DAIFMT_CBS_CFM:
329 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
330 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
331 IMX_AUDMUX_V2_PTCR_RFSDIR |
332 IMX_AUDMUX_V2_PTCR_TFSDIR;
333 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
334 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
335 IMX_AUDMUX_V2_PTCR_RCLKDIR |
336 IMX_AUDMUX_V2_PTCR_TCLKDIR;
338 case SND_SOC_DAIFMT_CBS_CFS:
339 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
340 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
341 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
342 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
343 IMX_AUDMUX_V2_PTCR_RFSDIR |
344 IMX_AUDMUX_V2_PTCR_RCLKDIR |
345 IMX_AUDMUX_V2_PTCR_TFSDIR |
346 IMX_AUDMUX_V2_PTCR_TCLKDIR;
352 /* Asynchronous mode can not be set along with RCLKDIR */
353 ret = imx_audmux_v2_configure_port(int_port, 0,
354 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
356 dev_err(dev, "audmux internal port setup failed\n");
360 ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
361 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
363 dev_err(dev, "audmux internal port setup failed\n");
367 ret = imx_audmux_v2_configure_port(ext_port, 0,
368 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
370 dev_err(dev, "audmux external port setup failed\n");
374 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
375 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
377 dev_err(dev, "audmux external port setup failed\n");
384 static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
386 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
387 struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
388 struct codec_priv *codec_priv = &priv->codec_priv;
389 struct device *dev = card->dev;
392 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
393 codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
395 dev_err(dev, "failed to set sysclk in %s\n", __func__);
402 static int fsl_asoc_card_probe(struct platform_device *pdev)
404 struct device_node *cpu_np, *codec_np, *asrc_np;
405 struct device_node *np = pdev->dev.of_node;
406 struct platform_device *asrc_pdev = NULL;
407 struct platform_device *cpu_pdev;
408 struct fsl_asoc_card_priv *priv;
409 struct i2c_client *codec_dev;
410 struct clk *codec_clk;
411 const char *codec_dai_name;
415 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
419 cpu_np = of_parse_phandle(np, "audio-cpu", 0);
420 /* Give a chance to old DT binding */
422 cpu_np = of_parse_phandle(np, "ssi-controller", 0);
423 codec_np = of_parse_phandle(np, "audio-codec", 0);
424 if (!cpu_np || !codec_np) {
425 dev_err(&pdev->dev, "phandle missing or invalid\n");
430 cpu_pdev = of_find_device_by_node(cpu_np);
432 dev_err(&pdev->dev, "failed to find CPU DAI device\n");
437 codec_dev = of_find_i2c_device_by_node(codec_np);
439 dev_err(&pdev->dev, "failed to find codec platform device\n");
444 asrc_np = of_parse_phandle(np, "audio-asrc", 0);
446 asrc_pdev = of_find_device_by_node(asrc_np);
448 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
449 codec_clk = clk_get(&codec_dev->dev, NULL);
450 if (!IS_ERR(codec_clk)) {
451 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
455 /* Default sample rate and format, will be updated in hw_params() */
456 priv->sample_rate = 44100;
457 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
459 /* Assign a default DAI format, and allow each card to overwrite it */
460 priv->dai_fmt = DAI_FMT_BASE;
462 /* Diversify the card configurations */
463 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
464 codec_dai_name = "cs42888";
465 priv->card.set_bias_level = NULL;
466 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
467 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
468 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
469 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
470 priv->cpu_priv.slot_width = 32;
471 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
472 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
473 codec_dai_name = "sgtl5000";
474 priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
475 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
476 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
477 codec_dai_name = "wm8962";
478 priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
479 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
480 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
481 priv->codec_priv.pll_id = WM8962_FLL;
482 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
483 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
484 codec_dai_name = "wm8960-hifi";
485 priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
486 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
487 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
488 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
490 dev_err(&pdev->dev, "unknown Device Tree compatible\n");
495 /* Common settings for corresponding Freescale CPU DAI driver */
496 if (strstr(cpu_np->name, "ssi")) {
497 /* Only SSI needs to configure AUDMUX */
498 ret = fsl_asoc_card_audmux_init(np, priv);
500 dev_err(&pdev->dev, "failed to init audmux\n");
503 } else if (strstr(cpu_np->name, "esai")) {
504 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
505 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
506 } else if (strstr(cpu_np->name, "sai")) {
507 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
508 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
511 sprintf(priv->name, "%s-audio", codec_dev->name);
513 /* Initialize sound card */
515 priv->card.dev = &pdev->dev;
516 priv->card.name = priv->name;
517 priv->card.dai_link = priv->dai_link;
518 priv->card.dapm_routes = audio_map;
519 priv->card.late_probe = fsl_asoc_card_late_probe;
520 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
521 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
522 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
524 memcpy(priv->dai_link, fsl_asoc_card_dai,
525 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
527 ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
529 dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
533 /* Normal DAI Link */
534 priv->dai_link[0].cpu_of_node = cpu_np;
535 priv->dai_link[0].codec_of_node = codec_np;
536 priv->dai_link[0].codec_dai_name = codec_dai_name;
537 priv->dai_link[0].platform_of_node = cpu_np;
538 priv->dai_link[0].dai_fmt = priv->dai_fmt;
539 priv->card.num_links = 1;
542 /* DPCM DAI Links only if ASRC exsits */
543 priv->dai_link[1].cpu_of_node = asrc_np;
544 priv->dai_link[1].platform_of_node = asrc_np;
545 priv->dai_link[2].codec_dai_name = codec_dai_name;
546 priv->dai_link[2].codec_of_node = codec_np;
547 priv->dai_link[2].cpu_of_node = cpu_np;
548 priv->dai_link[2].dai_fmt = priv->dai_fmt;
549 priv->card.num_links = 3;
551 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
554 dev_err(&pdev->dev, "failed to get output rate\n");
559 ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
561 dev_err(&pdev->dev, "failed to get output rate\n");
567 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
569 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
572 /* Finish card registering */
573 platform_set_drvdata(pdev, priv);
574 snd_soc_card_set_drvdata(&priv->card, priv);
576 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
578 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
581 of_node_put(asrc_np);
583 of_node_put(codec_np);
589 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
590 { .compatible = "fsl,imx-audio-cs42888", },
591 { .compatible = "fsl,imx-audio-sgtl5000", },
592 { .compatible = "fsl,imx-audio-wm8962", },
593 { .compatible = "fsl,imx-audio-wm8960", },
596 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
598 static struct platform_driver fsl_asoc_card_driver = {
599 .probe = fsl_asoc_card_probe,
601 .name = "fsl-asoc-card",
602 .pm = &snd_soc_pm_ops,
603 .of_match_table = fsl_asoc_card_dt_ids,
606 module_platform_driver(fsl_asoc_card_driver);
608 MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
609 MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
610 MODULE_ALIAS("platform:fsl-asoc-card");
611 MODULE_LICENSE("GPL");