/* * sound\soc\sunxi\sunxi_snddaudio0.c * (C) Copyright 2014-2016 * Reuuimlla Technology Co., Ltd. * huangxin * * some simple description for this code * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License as * published by the Free Software Foundation; either version 2 of * the License, or (at your option) any later version. * */#include #include #include #include #include #include #include #include #include #include "../../codecs/ac100.h" static int daudio_pcm_select; static int daudio_master; static int audio_format; static int signal_inversion; static int analog_bb; static int digital_bb; struct val_str { int *val; char *str; }; static struct val_str properties[] = { {&daudio_pcm_select, "daudio_select"}, {&daudio_master, "daudio_master"}, {&audio_format, "audio_format"}, {&signal_inversion, "signal_inversion"}, {&analog_bb, "analog_bb"}, {&digital_bb, "digital_bb"} }; static int sunxi_snddaudio0_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 *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int freq, clk_div; int ret; switch (params_rate(params)) { case 8000: case 16000: case 32000: case 64000: case 128000: case 12000: case 24000: case 48000: case 96000: case 192000: freq = 24576000; break; case 11025: case 22050: case 44100: case 88200: case 176400: freq = 22579200; break; default: pr_err("unsupport params rate\n"); return -EINVAL; } /* set the codec FLL */ #if defined (CONFIG_SND_SOC_AD82584F) ret = snd_soc_dai_set_pll(codec_dai, AC100_BCLK1, 0, 3072000, freq); #else ret = snd_soc_dai_set_pll(codec_dai, AC100_BCLK1, 0, 1024000, freq); #endif if (ret < 0) { pr_warn("[daudio0],the codec_dai set_pll failed.\n"); return ret; } /*set cpu_dai clk*/ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, freq, daudio_pcm_select); if (ret < 0) { pr_warn("[daudio0],the cpu_dai set_sysclk failed.\n"); return ret; } /*set codec_dai clk*/ ret = snd_soc_dai_set_sysclk(codec_dai, AIF1_CLK, freq, daudio_pcm_select); if (ret < 0) { pr_warn("[daudio0],the codec_dai set_sysclk failed.\n"); return ret; } /* * ac100: master. AP: slave */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { pr_warn("[daudio0],the codec_dai set_fmt failed.\n"); return ret; } ret = snd_soc_dai_set_fmt(cpu_dai, (audio_format | (signal_inversion<<8) | SND_SOC_DAIFMT_CBS_CFS)); if (ret < 0) { pr_warn("[daudio0],the cpu_dai set_format failed.\n"); return ret; } clk_div = freq / params_rate(params); ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, clk_div); if (ret < 0) { pr_warn("[daudio0],the cpu_dai set_clkdiv failed.\n"); return ret; } ret = snd_soc_dai_set_clkdiv(codec_dai, 0, clk_div); if (ret < 0) { pr_warn("[daudio0],the codec_dai set set_clkdiv failed.\n"); return ret; } return 0; } static int bt_net_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 *codec_dai = rtd->codec_dai; int ret = 0; int freq_in = 24576000; if (params_rate(params) != 8000) return -EINVAL; /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, AC100_MCLK1, 0, freq_in, freq_in); if (ret < 0) return ret; /*set codec system clock source aif2*/ ret = snd_soc_dai_set_sysclk(codec_dai, AIF2_CLK, 0, 0); if (ret < 0) return ret; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; return 0; } static int bb_voice_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 *codec_dai = rtd->codec_dai; int ret = 0; int freq_in = 24576000; if (analog_bb) { /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, AC100_MCLK1, 0, freq_in, freq_in); if (ret < 0) return ret; /*set codec system clock source aif2*/ ret = snd_soc_dai_set_sysclk(codec_dai, AIF2_CLK, 0, 0); if (ret < 0) return ret; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; } else if (digital_bb) { /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, AC100_BCLK2, 0, 2048000, freq_in); if (ret < 0) return ret; /*set system clock source aif2*/ ret = snd_soc_dai_set_sysclk(codec_dai, AIF2_CLK, 0, 0); if (ret < 0) return ret; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; } else { /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; } return 0; } static int phone_system_voice_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 *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret = 0; u32 freq_in = 22579200; unsigned long sample_rate = params_rate(params); switch (sample_rate) { case 8000: case 16000: case 32000: case 64000: case 128000: case 12000: case 24000: case 48000: case 96000: case 192000: freq_in = 24576000; break; } if (analog_bb) { /* set the codec FLL */ ret = snd_soc_dai_set_pll(codec_dai, AC100_MCLK1, 0, freq_in, freq_in); if (ret < 0) return ret; /*set cpu clk*/ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, freq_in, daudio_pcm_select); if (ret < 0) return ret; /*set codec clk*/ ret = snd_soc_dai_set_sysclk(codec_dai, AIF1_CLK, freq_in, daudio_pcm_select); if (ret < 0) return ret; /*ac100: master. AP: slave*/ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; ret = snd_soc_dai_set_fmt(cpu_dai, (audio_format | (signal_inversion<<8) | SND_SOC_DAIFMT_CBS_CFS)); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, sample_rate); if (ret < 0) return ret; } else if (digital_bb) { /*set cpu clk*/ ret = snd_soc_dai_set_sysclk(cpu_dai, 0, freq_in, daudio_pcm_select); if (ret < 0) return ret; /* ac100: master. AP: slave*/ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; ret = snd_soc_dai_set_fmt(cpu_dai, (audio_format | (signal_inversion<<8) | SND_SOC_DAIFMT_CBS_CFS)); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, sample_rate); if (ret < 0) return ret; } else { } return 0; } static int bt_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 *codec_dai = rtd->codec_dai; int ret = 0; if (params_rate(params) != 8000) return -EINVAL; /* set codec aif3 configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; return 0; } /* * Card initialization */ static int sunxi_daudio_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); pr_debug("%s,line:%d\n", __func__, __LINE__); snd_soc_dapm_disable_pin(dapm, "HPOUTR"); snd_soc_dapm_disable_pin(dapm, "HPOUTL"); snd_soc_dapm_disable_pin(dapm, "EAROUTP"); snd_soc_dapm_disable_pin(dapm, "EAROUTN"); snd_soc_dapm_disable_pin(dapm, "SPK1P"); snd_soc_dapm_disable_pin(dapm, "SPK2P"); snd_soc_dapm_disable_pin(dapm, "SPK1N"); snd_soc_dapm_disable_pin(dapm, "SPK2N"); snd_soc_dapm_disable_pin(dapm, "LINEOUTP"); snd_soc_dapm_disable_pin(dapm, "LINEOUTN"); snd_soc_dapm_disable_pin(dapm, "MIC1P"); snd_soc_dapm_disable_pin(dapm, "MIC1N"); snd_soc_dapm_disable_pin(dapm, "MIC2"); snd_soc_dapm_disable_pin(dapm, "MIC3"); snd_soc_dapm_disable_pin(dapm, "D_MIC"); snd_soc_dapm_disable_pin(&rtd->card->dapm, "External Speaker"); snd_soc_dapm_disable_pin(&rtd->card->dapm, "Headphone"); snd_soc_dapm_disable_pin(&rtd->card->dapm, "Earpiece"); snd_soc_dapm_disable_pin(&rtd->card->dapm, "Lineout"); snd_soc_dapm_sync(dapm); return 0; } static struct snd_soc_ops sunxi_snddaudio_ops = { .hw_params = sunxi_snddaudio0_hw_params, }; static struct snd_soc_ops bt_net_ops = { .hw_params = bt_net_hw_params, }; static struct snd_soc_ops bb_voice_ops = { .hw_params = bb_voice_hw_params, }; static struct snd_soc_ops phone_system_voice_ops = { .hw_params = phone_system_voice_hw_params, }; static struct snd_soc_ops bt_voice_ops = { .hw_params = bt_hw_params, }; static const struct snd_kcontrol_new ac100_pin_controls[] = { SOC_DAPM_PIN_SWITCH("External Speaker"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Earpiece"), SOC_DAPM_PIN_SWITCH("Lineout"), }; static const struct snd_soc_dapm_widget a80_ac100_dapm_widgets[] = { SND_SOC_DAPM_MIC("External MainMic", NULL), SND_SOC_DAPM_MIC("HeadphoneMic", NULL), SND_SOC_DAPM_MIC("DigitalMic", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { {"MainMic Bias", NULL, "External MainMic"}, {"MIC1P", NULL, "MainMic Bias"}, {"MIC1N", NULL, "MainMic Bias"}, {"MIC2", NULL, "HMic Bias"}, {"MIC3", NULL, "MainMic Bias"}, {"HMic Bias", NULL, "HeadphoneMic"}, /*d-mic*/ {"MainMic Bias", NULL, "DigitalMic"}, {"D_MIC", NULL, "MainMic Bias"}, }; static struct snd_soc_dai_link sunxi_snddaudio_dai_link[] = { { .name = "s_i2s1", .cpu_dai_name = "sunxi-tdm", .stream_name = "SUNXI-CODEC", .codec_dai_name = "ac100-aif1", .codec_name = "ac100-codec", .init = sunxi_daudio_init, .ops = &sunxi_snddaudio_ops, }, {/* Second DAI i/f */ .name = "ac100 Voice", .stream_name = "Voice", .cpu_dai_name = "bb-voice-dai", .codec_dai_name = "ac100-aif2", .codec_name = "ac100-codec", .ops = &bb_voice_ops, }, {/*phone cap and keytone*/ .name = "VIR", .cpu_dai_name = "sec_dai", .stream_name = "vir-dai", .codec_dai_name = "ac100-aif1", .codec_name = "ac100-codec", .ops = &phone_system_voice_ops, }, { /*bt*/ .name = "BT", .cpu_dai_name = "bb-voice-dai", .stream_name = "bt-dai", .codec_dai_name = "ac100-aif3", .codec_name = "ac100-codec", .ops = &bt_voice_ops, }, { /*bt-network*/ .name = "BT-NET", .cpu_dai_name = "bb-voice-dai", .stream_name = "aif2-bt-net", .codec_dai_name = "ac100-aif2", .codec_name = "ac100-codec", .ops = &bt_net_ops, #if defined (CONFIG_SND_SOC_AD82584F) }, { .name = "digital PA", .cpu_dai_name = "sunxi-tdm", .stream_name = "digital-codec", .codec_dai_name = "ad82584f", .codec_name = "ad82584f.1-0031", #endif } }; static struct snd_soc_card snd_soc_sunxi_snddaudio = { .name = "snddaudio0", .owner = THIS_MODULE, .dai_link = sunxi_snddaudio_dai_link, .num_links = ARRAY_SIZE(sunxi_snddaudio_dai_link), .dapm_widgets = a80_ac100_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(a80_ac100_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .controls = ac100_pin_controls, .num_controls = ARRAY_SIZE(ac100_pin_controls), }; static int sunxi_snddaudio0_dev_probe(struct platform_device *pdev) { int ret = 0; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_sunxi_snddaudio; unsigned int val; unsigned int i; card->dev = &pdev->dev; dev_warn(&pdev->dev, "%s,line:%d\n", __func__, __LINE__); for (i = 0; i < ARRAY_SIZE(properties); i++) { ret = of_property_read_u32(np, properties[i].str, &val); if (ret < 0) { dev_warn(&pdev->dev, "%s config missing or invalid\n", properties[i].str); *(properties[i].val) = 0; } else { *(properties[i].val) = val; pr_debug("%s=%d\n", properties[i].str, *(properties[i].val)); } } dev_warn(&pdev->dev, "%s,line:%d\n", __func__, __LINE__); sunxi_snddaudio_dai_link[0].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[0].cpu_of_node = of_parse_phandle(np, "sunxi,daudio0-controller", 0); if (!sunxi_snddaudio_dai_link[0].cpu_of_node) { dev_err(&pdev->dev, "Property 'sunxi,daudio0-controller' missing\n"); ret = -EINVAL; } sunxi_snddaudio_dai_link[0].platform_name = NULL; sunxi_snddaudio_dai_link[0].platform_of_node = sunxi_snddaudio_dai_link[0].cpu_of_node; #if defined(CONFIG_SND_SOC_AD82584F) sunxi_snddaudio_dai_link[5].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[5].cpu_of_node = of_parse_phandle(np, "sunxi,daudio0-controller", 0); if (!sunxi_snddaudio_dai_link[5].cpu_of_node) { dev_err(&pdev->dev, "Property 'sunxi,daudio0-controller' missing\n"); ret = -EINVAL; } sunxi_snddaudio_dai_link[5].platform_name = NULL; sunxi_snddaudio_dai_link[5].platform_of_node = sunxi_snddaudio_dai_link[5].cpu_of_node; #endif #if 1 sunxi_snddaudio_dai_link[1].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[1].cpu_of_node = of_parse_phandle(np, "sunxi,bbdai-controller", 0); if (!sunxi_snddaudio_dai_link[1].cpu_of_node) { dev_err(&pdev->dev, "Property 'sunxi,bbdai-controller' missing\n"); // ret = -EINVAL; } sunxi_snddaudio_dai_link[2].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[2].cpu_of_node = sunxi_snddaudio_dai_link[1].cpu_of_node; sunxi_snddaudio_dai_link[3].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[3].cpu_of_node = sunxi_snddaudio_dai_link[1].cpu_of_node; sunxi_snddaudio_dai_link[4].cpu_dai_name = NULL; sunxi_snddaudio_dai_link[4].cpu_of_node = sunxi_snddaudio_dai_link[1].cpu_of_node; #endif /* codec_of_node */ /* sunxi_snddaudio_dai_link[0].codec_name = NULL; sunxi_snddaudio_dai_link[0].codec_of_node = of_find_compatible_node(NULL, NULL, "allwinner,sunxi-ac100-codec"); if (!sunxi_snddaudio_dai_link[0].codec_of_node) { dev_err(&pdev->dev, "codec_of_node missing or invalid\n"); ret = -EINVAL; } */ #if 0 for (i = 1; i < ARRAY_SIZE(sunxi_snddaudio_dai_link); i++) { sunxi_snddaudio_dai_link[i].codec_name = NULL; sunxi_snddaudio_dai_link[i].codec_of_node = sunxi_snddaudio_dai_link[0].codec_of_node; } #endif dev_warn(&pdev->dev, "%s,line:%d\n", __func__, __LINE__); ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); } dev_warn(&pdev->dev, "%s,line:%d\n", __func__, __LINE__); return ret; } static int __exit sunxi_snddaudio0_dev_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static const struct of_device_id sunxi_daudio0_of_match[] = { { .compatible = "allwinner,sunxi-daudio0-machine", }, {}, }; /*method relating*/ static struct platform_driver sunxi_daudio_driver = { .driver = { .name = "snddaudio0", .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = sunxi_daudio0_of_match, }, .probe = sunxi_snddaudio0_dev_probe, .remove = __exit_p(sunxi_snddaudio0_dev_remove), }; module_platform_driver(sunxi_daudio_driver); MODULE_AUTHOR("huangxin"); MODULE_DESCRIPTION("SUNXI_snddaudio ALSA SoC audio driver"); MODULE_LICENSE("GPL");