/* * Copyright (C) 2010-2016 Freescale Semiconductor, Inc. All Rights Reserved. */ /* * The code contained herein is licensed under the GNU General Public * License. You may obtain a copy of the GNU General Public License * Version 2 or later at the following locations: * * http://www.opensource.org/licenses/gpl-license.html * http://www.gnu.org/copyleft/gpl.html */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include "fsl_esai.h" #define CODEC_CLK_EXTER_OSC 1 #define CODEC_CLK_ESAI_HCKT 2 #define SUPPORT_RATE_NUM 10 struct imx_priv { struct clk *codec_clk; struct clk *esai_clk; unsigned int mclk_freq; unsigned int esai_freq; struct platform_device *pdev; struct platform_device *asrc_pdev; u32 asrc_rate; u32 asrc_format; bool is_codec_master; bool is_codec_rpmsg; bool is_stream_in_use[2]; bool is_stream_tdm[2]; }; static struct imx_priv card_priv; static int imx_cs42888_surround_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct imx_priv *priv = &card_priv; struct device *dev = &priv->pdev->dev; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 channels = params_channels(params); u32 max_tdm_rate; u32 dai_format; int ret = 0; priv->is_stream_tdm[tx] = channels > 1 && channels % 2; dai_format = SND_SOC_DAIFMT_NB_NF | (priv->is_stream_tdm[tx] ? SND_SOC_DAIFMT_DSP_A : SND_SOC_DAIFMT_LEFT_J); priv->is_stream_in_use[tx] = true; if (priv->is_stream_in_use[!tx] && (priv->is_stream_tdm[tx] != priv->is_stream_tdm[!tx])) { dev_err(dev, "Don't support different fmt for tx & rx\n"); return -EINVAL; } priv->mclk_freq = clk_get_rate(priv->codec_clk); priv->esai_freq = clk_get_rate(priv->esai_clk); if (priv->is_codec_master) { /* TDM is not supported by codec in master mode */ if (priv->is_stream_tdm[tx]) { dev_err(dev, "%d channels are not supported in codec master mode\n", channels); return -EINVAL; } dai_format |= SND_SOC_DAIFMT_CBM_CFM; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL, priv->mclk_freq, SND_SOC_CLOCK_IN); else ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL, priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "failed to set cpu sysclk: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, priv->mclk_freq, SND_SOC_CLOCK_OUT); if (ret) { dev_err(dev, "failed to set codec sysclk: %d\n", ret); return ret; } } else { dai_format |= SND_SOC_DAIFMT_CBS_CFS; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL, priv->mclk_freq, SND_SOC_CLOCK_OUT); else ret = snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL, priv->mclk_freq, SND_SOC_CLOCK_OUT); if (ret) { dev_err(dev, "failed to set cpu sysclk: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { dev_err(dev, "failed to set codec sysclk: %d\n", ret); return ret; } } /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); if (ret) { dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); return ret; } /* set i.MX active slot mask */ if (priv->is_stream_tdm[tx]) { /* 2 required by ESAI BCLK divisors, 8 slots, 32 width */ if (priv->is_codec_master) max_tdm_rate = priv->mclk_freq / (8*32); else max_tdm_rate = priv->esai_freq / (2*8*32); if (params_rate(params) > max_tdm_rate) { dev_err(dev, "maximum supported sampling rate for %d channels is %dKHz\n", channels, max_tdm_rate / 1000); return -EINVAL; } /* * Per datasheet, the codec expects 8 slots and 32 bits * for every slot in TDM mode. */ snd_soc_dai_set_tdm_slot(cpu_dai, BIT(channels) - 1, BIT(channels) - 1, 8, 32); } else snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, dai_format); if (ret) { dev_err(dev, "failed to set codec dai fmt: %d\n", ret); return ret; } return 0; } static int imx_cs42888_surround_hw_free(struct snd_pcm_substream *substream) { struct imx_priv *priv = &card_priv; bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; priv->is_stream_in_use[tx] = false; return 0; } static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; static struct snd_pcm_hw_constraint_list constraint_rates; struct imx_priv *priv = &card_priv; struct device *dev = &priv->pdev->dev; static u32 support_rates[SUPPORT_RATE_NUM]; int ret; priv->mclk_freq = clk_get_rate(priv->codec_clk); if (priv->mclk_freq % 12288000 == 0) { support_rates[0] = 48000; support_rates[1] = 96000; support_rates[2] = 192000; constraint_rates.list = support_rates; constraint_rates.count = 3; ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraint_rates); if (ret) return ret; } else dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq); return 0; } static struct snd_soc_ops imx_cs42888_surround_ops = { .startup = imx_cs42888_surround_startup, .hw_params = imx_cs42888_surround_hw_params, .hw_free = imx_cs42888_surround_hw_free, }; /** * imx_cs42888_surround_startup() is to set constrain for hw parameter, but * backend use same runtime as frontend, for p2p backend need to use different * parameter, so backend can't use the startup. */ static struct snd_soc_ops imx_cs42888_surround_ops_be = { .hw_params = imx_cs42888_surround_hw_params, .hw_free = imx_cs42888_surround_hw_free, }; static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), SND_SOC_DAPM_LINE("Line In Jack", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { /* Line out jack */ {"Line Out Jack", NULL, "AOUT1L"}, {"Line Out Jack", NULL, "AOUT1R"}, {"Line Out Jack", NULL, "AOUT2L"}, {"Line Out Jack", NULL, "AOUT2R"}, {"Line Out Jack", NULL, "AOUT3L"}, {"Line Out Jack", NULL, "AOUT3R"}, {"Line Out Jack", NULL, "AOUT4L"}, {"Line Out Jack", NULL, "AOUT4R"}, {"AIN1L", NULL, "Line In Jack"}, {"AIN1R", NULL, "Line In Jack"}, {"AIN2L", NULL, "Line In Jack"}, {"AIN2R", NULL, "Line In Jack"}, {"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */ {"CPU-Capture", NULL, "Capture"}, {"CPU-Playback", NULL, "ASRC-Playback"}, {"ASRC-Capture", NULL, "CPU-Capture"}, }; static const struct snd_soc_dapm_route audio_map_v2[] = { /* Line out jack */ {"Line Out Jack", NULL, "AOUT1L"}, {"Line Out Jack", NULL, "AOUT1R"}, {"Line Out Jack", NULL, "AOUT2L"}, {"Line Out Jack", NULL, "AOUT2R"}, {"Line Out Jack", NULL, "AOUT3L"}, {"Line Out Jack", NULL, "AOUT3R"}, {"Line Out Jack", NULL, "AOUT4L"}, {"Line Out Jack", NULL, "AOUT4R"}, {"AIN1L", NULL, "Line In Jack"}, {"AIN1R", NULL, "Line In Jack"}, {"AIN2L", NULL, "Line In Jack"}, {"AIN2R", NULL, "Line In Jack"}, {"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */ {"CPU-Capture", NULL, "Capture"}, {"CPU-Playback", NULL, "CLIENT0-Playback"}, {"CLIENT0-Capture", NULL, "CPU-Capture"}, {"CPU-Playback", NULL, "CLIENT1-Playback"}, {"CLIENT1-Capture", NULL, "CPU-Capture"}, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct imx_priv *priv = &card_priv; struct snd_interval *rate; struct snd_mask *mask; if (!priv->asrc_pdev) return -EINVAL; rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); rate->max = rate->min = priv->asrc_rate; mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(mask); snd_mask_set(mask, priv->asrc_format); return 0; } static int be_hw_params_fixup_v2(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(rtd, SNDRV_PCM_STREAM_PLAYBACK); struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct fsl_esai *esai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct fsl_esai_mix *mix = &esai->mix[tx]; struct snd_interval *channels; channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels->max = channels->min = mix->channels; return 0; } SND_SOC_DAILINK_DEFS(hifi, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "cs42888")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_fe2, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "cs42888")), DAILINK_COMP_ARRAY(COMP_DUMMY())); static struct snd_soc_dai_link imx_cs42888_dai[] = { { .name = "HiFi", .stream_name = "HiFi", .ops = &imx_cs42888_surround_ops, .ignore_pmdown_time = 1, SND_SOC_DAILINK_REG(hifi), }, { .name = "HiFi-ASRC-FE", .stream_name = "HiFi-ASRC-FE", .dynamic = 1, .ignore_pmdown_time = 1, .dpcm_playback = 1, .dpcm_capture = 1, .dpcm_merged_chan = 1, SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-BE", .stream_name = "HiFi-ASRC-BE", .no_pcm = 1, .ignore_pmdown_time = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &imx_cs42888_surround_ops_be, .be_hw_params_fixup = be_hw_params_fixup, SND_SOC_DAILINK_REG(hifi_be), }, }; static struct snd_soc_dai_link imx_cs42888_dai_v2[] = { { .name = "HiFi-ASRC-FE1", .stream_name = "HiFi-ASRC-FE1", .dynamic = 1, .ignore_pmdown_time = 1, .dpcm_playback = 1, .dpcm_capture = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-FE2", .stream_name = "HiFi-ASRC-FE2", .dynamic = 1, .ignore_pmdown_time = 1, .dpcm_playback = 1, .dpcm_capture = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(hifi_fe2), }, { .name = "HiFi-ASRC-BE", .stream_name = "HiFi-ASRC-BE", .no_pcm = 1, .ignore_pmdown_time = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &imx_cs42888_surround_ops_be, .be_hw_params_fixup = be_hw_params_fixup_v2, SND_SOC_DAILINK_REG(hifi_be), }, }; static struct snd_soc_card snd_soc_card_imx_cs42888 = { .name = "cs42888-audio", .dai_link = imx_cs42888_dai, .dapm_widgets = imx_cs42888_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .owner = THIS_MODULE, }; /* * This function will register the snd_soc_pcm_link drivers. */ static int imx_cs42888_probe(struct platform_device *pdev) { struct device_node *esai_np, *codec_np; struct device_node *asrc_np = NULL; struct platform_device *esai_pdev; struct platform_device *asrc_pdev = NULL; struct imx_priv *priv = &card_priv; struct of_phandle_args args[2]; struct platform_device *cpu_pdev[2]; int ret; u32 width; priv->pdev = pdev; priv->asrc_pdev = NULL; if (of_property_read_bool(pdev->dev.of_node, "codec-rpmsg")) priv->is_codec_rpmsg = true; esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0); codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); if (!esai_np || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); ret = -EINVAL; goto fail; } asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0); if (asrc_np) { asrc_pdev = of_find_device_by_node(asrc_np); priv->asrc_pdev = asrc_pdev; } esai_pdev = of_find_device_by_node(esai_np); if (!esai_pdev) { dev_err(&pdev->dev, "failed to find ESAI platform device\n"); ret = -EINVAL; goto fail; } if (priv->is_codec_rpmsg) { struct platform_device *codec_dev; codec_dev = of_find_device_by_node(codec_np); if (!codec_dev || !codec_dev->dev.driver) { dev_err(&pdev->dev, "failed to find codec platform device\n"); ret = -EINVAL; goto fail; } priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(priv->codec_clk)) { ret = PTR_ERR(priv->codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } } else { struct i2c_client *codec_dev; codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev || !codec_dev->dev.driver) { dev_dbg(&pdev->dev, "failed to find codec platform device\n"); ret = -EPROBE_DEFER; goto fail; } priv->codec_clk = devm_clk_get(&codec_dev->dev, NULL); if (IS_ERR(priv->codec_clk)) { ret = PTR_ERR(priv->codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } } if (priv->is_codec_rpmsg) { imx_cs42888_dai[0].codecs->name = "rpmsg-audio-codec-cs42888"; imx_cs42888_dai[0].codecs->dai_name = "cs42888"; } else { imx_cs42888_dai[0].codecs->of_node = codec_np; } /*if there is no asrc controller, we only enable one device*/ if (!asrc_pdev) { imx_cs42888_dai[0].cpus->dai_name = dev_name(&esai_pdev->dev); imx_cs42888_dai[0].platforms->of_node = esai_np; snd_soc_card_imx_cs42888.num_links = 1; snd_soc_card_imx_cs42888.num_dapm_routes = ARRAY_SIZE(audio_map) - 2; } else { imx_cs42888_dai[0].cpus->dai_name = dev_name(&esai_pdev->dev); imx_cs42888_dai[0].platforms->of_node = esai_np; imx_cs42888_dai[1].cpus->of_node = asrc_np; imx_cs42888_dai[1].platforms->of_node = asrc_np; imx_cs42888_dai[2].cpus->dai_name = dev_name(&esai_pdev->dev); snd_soc_card_imx_cs42888.num_links = 3; if (priv->is_codec_rpmsg) { imx_cs42888_dai[2].codecs->name = "rpmsg-audio-codec-cs42888"; imx_cs42888_dai[2].codecs->dai_name = "cs42888"; } else { imx_cs42888_dai[2].codecs->of_node = codec_np; } ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", &priv->asrc_rate); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); ret = -EINVAL; goto fail; } ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); ret = -EINVAL; goto fail; } if (width == 24) priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE; else priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE; } /* switch to v2 if there is "client-dais" property */ if (of_property_read_bool(esai_np, "client-dais")) { ret = of_parse_phandle_with_args(esai_np, "client-dais", NULL, 0, &args[0]); if (ret < 0) { dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); return ret; } cpu_pdev[0] = of_find_device_by_node(args[0].np); if (!cpu_pdev[0]) { dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } ret = of_parse_phandle_with_args(esai_np, "client-dais", NULL, 1, &args[1]); if (ret < 0) { dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); return ret; } cpu_pdev[1] = of_find_device_by_node(args[1].np); if (!cpu_pdev[1]) { dev_err(&pdev->dev, "failed to find SAI platform device\n"); return -EINVAL; } imx_cs42888_dai_v2[0].cpus->dai_name = dev_name(&cpu_pdev[0]->dev); imx_cs42888_dai_v2[0].cpus->of_node = args[0].np; imx_cs42888_dai_v2[0].platforms->of_node = args[0].np; imx_cs42888_dai_v2[1].cpus->dai_name = dev_name(&cpu_pdev[1]->dev); imx_cs42888_dai_v2[1].cpus->of_node = args[1].np; imx_cs42888_dai_v2[1].platforms->of_node = args[1].np; imx_cs42888_dai_v2[2].cpus->dai_name = dev_name(&esai_pdev->dev); imx_cs42888_dai_v2[2].codecs->of_node = codec_np; snd_soc_card_imx_cs42888.dai_link = imx_cs42888_dai_v2; snd_soc_card_imx_cs42888.num_links = 3; snd_soc_card_imx_cs42888.dapm_routes = audio_map_v2, snd_soc_card_imx_cs42888.num_dapm_routes = ARRAY_SIZE(audio_map_v2); } priv->esai_clk = devm_clk_get(&esai_pdev->dev, "extal"); if (IS_ERR(priv->esai_clk)) { ret = PTR_ERR(priv->esai_clk); dev_err(&esai_pdev->dev, "failed to get cpu clk: %d\n", ret); goto fail; } priv->is_codec_master = false; if (of_property_read_bool(pdev->dev.of_node, "codec-master")) priv->is_codec_master = true; snd_soc_card_imx_cs42888.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&snd_soc_card_imx_cs42888, "model"); if (ret) goto fail; platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888); ret = snd_soc_register_card(&snd_soc_card_imx_cs42888); if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); fail: if (asrc_np) of_node_put(asrc_np); if (esai_np) of_node_put(esai_np); if (codec_np) of_node_put(codec_np); return ret; } static int imx_cs42888_remove(struct platform_device *pdev) { snd_soc_unregister_card(&snd_soc_card_imx_cs42888); return 0; } static const struct of_device_id imx_cs42888_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { /* sentinel */ } }; static struct platform_driver imx_cs42888_driver = { .probe = imx_cs42888_probe, .remove = imx_cs42888_remove, .driver = { .name = "imx-cs42888", .pm = &snd_soc_pm_ops, .of_match_table = imx_cs42888_dt_ids, }, }; module_platform_driver(imx_cs42888_driver); MODULE_AUTHOR("Freescale Semiconductor, Inc."); MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver"); MODULE_ALIAS("platform:imx-cs42888"); MODULE_LICENSE("GPL");