/* * sdp4430.c -- SoC audio for TI OMAP4430 SDP * * Author: Misael Lopez Cruz * Liam Girdwood * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "omap-mcpdm.h" #include "omap-abe.h" #include "omap-abe-dsp.h" #include "omap-pcm.h" #include "omap-mcbsp.h" #include "omap-dmic.h" #include "../codecs/twl6040.h" static struct regulator *av_switch_reg; static int twl6040_power_mode; static int mcbsp_cfg; static struct snd_soc_codec *twl6040_codec; static int sdp4430_modem_mcbsp_configure(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, int flag) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_substream *modem_substream[2]; struct snd_soc_pcm_runtime *modem_rtd; int channels; if (flag) { modem_substream[substream->stream] = snd_soc_get_dai_substream(rtd->card, OMAP_ABE_BE_MM_EXT1, substream->stream); if (unlikely(modem_substream[substream->stream] == NULL)) return -ENODEV; modem_rtd = modem_substream[substream->stream]->private_data; if (!mcbsp_cfg) { /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(modem_rtd->cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (unlikely(ret < 0)) { printk(KERN_ERR "can't set Modem cpu DAI configuration\n"); goto exit; } else { mcbsp_cfg = 1; } } if (params != NULL) { /* Configure McBSP internal buffer usage */ /* this need to be done for playback and/or record */ channels = params_channels(params); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) omap_mcbsp_set_rx_threshold( modem_rtd->cpu_dai->id, channels); else omap_mcbsp_set_tx_threshold( modem_rtd->cpu_dai->id, channels); } } else { mcbsp_cfg = 0; } exit: return ret; } static int sdp4430_modem_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { int ret; ret = sdp4430_modem_mcbsp_configure(substream, params, 1); if (ret) printk(KERN_ERR "can't set modem cpu DAI configuration\n"); return ret; } static int sdp4430_modem_hw_free(struct snd_pcm_substream *substream) { int ret; ret = sdp4430_modem_mcbsp_configure(substream, NULL, 0); if (ret) printk(KERN_ERR "can't clear modem cpu DAI configuration\n"); return ret; } static struct snd_soc_ops sdp4430_modem_ops = { .hw_params = sdp4430_modem_hw_params, .hw_free = sdp4430_modem_hw_free, }; static int sdp4430_mcpdm_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct twl6040 *twl6040 = codec->control_data; int clk_id, freq, ret; /* TWL6040 supplies McPDM PAD_CLKS */ ret = twl6040_enable(twl6040); if (ret) { printk(KERN_ERR "failed to enable TWL6040\n"); return ret; } if (twl6040_power_mode) { clk_id = TWL6040_HPPLL_ID; freq = 38400000; /* * TWL6040 requires MCLK to be active as long as * high-performance mode is in use. Glitch-free mux * cannot tolerate MCLK gating */ ret = cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 1); if (ret) { printk(KERN_ERR "failed to enable twl6040 MCLK\n"); goto err; } } else { clk_id = TWL6040_LPPLL_ID; freq = 32768; } /* set the codec mclk */ ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, SND_SOC_CLOCK_IN); if (ret) { printk(KERN_ERR "can't set codec system clock\n"); goto err; } /* low-power mode uses 32k clock, MCLK is not required */ if (!twl6040_power_mode) { ret = cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); if (ret) printk(KERN_ERR "failed to disable twl6040 MCLK\n"); } return 0; err: twl6040_disable(twl6040); return ret; } static void sdp4430_mcpdm_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct twl6040 *twl6040 = codec->control_data; /* TWL6040 supplies McPDM PAD_CLKS */ twl6040_disable(twl6040); } static struct snd_soc_ops sdp4430_mcpdm_ops = { .startup = sdp4430_mcpdm_startup, .shutdown = sdp4430_mcpdm_shutdown, }; static int sdp4430_mcbsp_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; int ret = 0; unsigned int channels; ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration\n"); return ret; } if (params != NULL) { /* Configure McBSP internal buffer usage */ /* this need to be done for playback and/or record */ channels = params_channels(params); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) omap_mcbsp_set_tx_threshold( cpu_dai->id, channels); else omap_mcbsp_set_rx_threshold( cpu_dai->id, channels); } /* * TODO: where does this clock come from (external source??) - * do we need to enable it. */ /* Set McBSP clock to external */ ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 64 * params_rate(params), SND_SOC_CLOCK_IN); if (ret < 0) printk(KERN_ERR "can't set cpu system clock\n"); return ret; } static struct snd_soc_ops sdp4430_mcbsp_ops = { .hw_params = sdp4430_mcbsp_hw_params, }; static int sdp4430_dmic_startup(struct snd_pcm_substream *substream) { struct twl6040 *twl6040 = twl6040_codec->control_data; /* In order for the DMIC's to use the PAD CLOCKS, the twl6040 * must be powered up, since it supplies the clock source. */ return twl6040_enable(twl6040); } static void sdp4430_dmic_shutdown(struct snd_pcm_substream *substream) { struct twl6040 *twl6040 = twl6040_codec->control_data; twl6040_disable(twl6040); } static int sdp4430_dmic_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; int ret = 0; if (!rtd->dai_link->no_pcm) ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS, 24000000, SND_SOC_CLOCK_IN); else ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, 19200000, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "can't set DMIC cpu system clock\n"); return ret; } if (!rtd->dai_link->no_pcm) ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_DMIC_CLKDIV, 10); else ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_DMIC_CLKDIV, 8); if (ret < 0) { printk(KERN_ERR "can't set DMIC cpu clock divider\n"); return ret; } return 0; } static struct snd_soc_ops sdp4430_dmic_ops = { .startup = sdp4430_dmic_startup, .shutdown = sdp4430_dmic_shutdown, .hw_params = sdp4430_dmic_hw_params, }; static int mcbsp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels->min = 2; snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - SNDRV_PCM_HW_PARAM_FIRST_MASK], SNDRV_PCM_FORMAT_S16_LE); return 0; } static int dmic_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); /* The ABE will covert the FE rate to 96k */ rate->min = rate->max = 96000; channels->min = channels->max = 2; snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - SNDRV_PCM_HW_PARAM_FIRST_MASK], SNDRV_PCM_FORMAT_S32_LE); return 0; } /* Headset jack */ static struct snd_soc_jack hs_jack; /*Headset jack detection DAPM pins */ static struct snd_soc_jack_pin hs_jack_pins[] = { { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, }; static int sdp4430_av_switch_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { int ret; if (SND_SOC_DAPM_EVENT_ON(event)) ret = regulator_enable(av_switch_reg); else ret = regulator_disable(av_switch_reg); return ret; } static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { ucontrol->value.integer.value[0] = twl6040_power_mode; return 0; } static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { if (twl6040_power_mode == ucontrol->value.integer.value[0]) return 0; twl6040_power_mode = ucontrol->value.integer.value[0]; abe_dsp_set_power_mode(twl6040_power_mode); return 1; } static const char *power_texts[] = {"Low-Power", "High-Performance"}; static const struct soc_enum sdp4430_enum[] = { SOC_ENUM_SINGLE_EXT(2, power_texts), }; static const struct snd_kcontrol_new sdp4430_controls[] = { SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0], sdp4430_get_power_mode, sdp4430_set_power_mode), }; /* SDP4430 machine DAPM */ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), SND_SOC_DAPM_SUPPLY("AV Switch Supply", SND_SOC_NOPM, 0, 0, sdp4430_av_switch_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIC("Digital Mic 0", NULL), SND_SOC_DAPM_MIC("Digital Mic 1", NULL), SND_SOC_DAPM_MIC("Digital Mic 2", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { /* External Mics: MAINMIC, SUBMIC with bias*/ {"MAINMIC", NULL, "Main Mic Bias"}, {"SUBMIC", NULL, "Main Mic Bias"}, {"Main Mic Bias", NULL, "Ext Mic"}, /* External Speakers: HFL, HFR */ {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, /* Headset Mic: HSMIC with bias */ {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Mic"}, {"Headset Mic", NULL, "AV Switch Supply"}, /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, {"Headset Stereophone", NULL, "HSOR"}, /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ {"AFML", NULL, "Aux/FM Stereo In"}, {"AFMR", NULL, "Aux/FM Stereo In"}, /* Digital Mics: DMic0, DMic1, DMic2 with bias */ {"DMIC0", NULL, "Digital Mic1 Bias"}, {"Digital Mic1 Bias", NULL, "Digital Mic 0"}, {"DMIC1", NULL, "Digital Mic1 Bias"}, {"Digital Mic1 Bias", NULL, "Digital Mic 1"}, {"DMIC2", NULL, "Digital Mic1 Bias"}, {"Digital Mic1 Bias", NULL, "Digital Mic 2"}, }; static int sdp4430_set_pdm_dl1_gains(struct snd_soc_dapm_context *dapm) { int output, val; if (snd_soc_dapm_get_pin_power(dapm, "Earphone Spk")) { output = OMAP_ABE_DL1_EARPIECE; } else if (snd_soc_dapm_get_pin_power(dapm, "Headset Stereophone")) { val = snd_soc_read(twl6040_codec, TWL6040_REG_HSLCTL); if (val & TWL6040_HSDACMODEL) /* HSDACL in LP mode */ output = OMAP_ABE_DL1_HEADSET_LP; else /* HSDACL in HP mode */ output = OMAP_ABE_DL1_HEADSET_HP; #if !defined(CONFIG_SND_OMAP_SOC_ABE_DL2) } else if (snd_soc_dapm_get_pin_power(dapm, "Ext Spk")) { output = OMAP_ABE_DL1_HANDSFREE; #endif } else { output = OMAP_ABE_DL1_NO_PDM; } return omap_abe_set_dl1_output(output); } static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct twl6040 *twl6040 = codec->control_data; struct snd_soc_dapm_context *dapm = &codec->dapm; int hsotrim, left_offset, right_offset, mode, ret; /* Add SDP4430 specific controls */ ret = snd_soc_add_controls(codec, sdp4430_controls, ARRAY_SIZE(sdp4430_controls)); if (ret) return ret; /* Add SDP4430 specific widgets */ ret = snd_soc_dapm_new_controls(dapm, sdp4430_twl6040_dapm_widgets, ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); if (ret) return ret; /* Set up SDP4430 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); snd_soc_dapm_enable_pin(dapm, "AFML"); snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); /* allow audio paths from the audio modem to run during suspend */ snd_soc_dapm_ignore_suspend(dapm, "Ext Mic"); snd_soc_dapm_ignore_suspend(dapm, "Ext Spk"); snd_soc_dapm_ignore_suspend(dapm, "AFML"); snd_soc_dapm_ignore_suspend(dapm, "AFMR"); snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(dapm, "Headset Stereophone"); snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 0"); snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 1"); snd_soc_dapm_ignore_suspend(dapm, "Digital Mic 2"); ret = snd_soc_dapm_sync(dapm); if (ret) return ret; /* Headset jack detection */ ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &hs_jack); if (ret) return ret; ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), hs_jack_pins); if (machine_is_omap_4430sdp() || machine_is_omap_tabletblaze() || machine_is_omap4_panda()) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); else snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); /* DC offset cancellation computation */ hsotrim = snd_soc_read(codec, TWL6040_REG_HSOTRIM); right_offset = (hsotrim & TWL6040_HSRO) >> TWL6040_HSRO_OFFSET; left_offset = hsotrim & TWL6040_HSLO; if (twl6040_get_icrev(twl6040) < TWL6040_REV_1_3) /* For ES under ES_1.3 HS step is 2 mV */ mode = 2; else /* For ES_1.3 HS step is 1 mV */ mode = 1; abe_dsp_set_hs_offset(left_offset, right_offset, mode); /* don't wait before switching of HS power */ rtd->pmdown_time = 0; return ret; } static int sdp4430_twl6040_dl2_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int hfotrim, left_offset, right_offset; /* DC offset cancellation computation */ hfotrim = snd_soc_read(codec, TWL6040_REG_HFOTRIM); right_offset = (hfotrim & TWL6040_HFRO) >> TWL6040_HFRO_OFFSET; left_offset = hfotrim & TWL6040_HFLO; abe_dsp_set_hf_offset(left_offset, right_offset); /* don't wait before switching of HF power */ rtd->pmdown_time = 0; return 0; } /* SDP4430 digital microphones DAPM */ static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { SND_SOC_DAPM_MIC("Digital Mic Legacy", NULL), }; static const struct snd_soc_dapm_route dmic_audio_map[] = { {"DMic", NULL, "Digital Mic1 Bias"}, {"Digital Mic1 Bias", NULL, "Digital Mic Legacy"}, }; static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); if (ret) return ret; ret = snd_soc_dapm_add_routes(dapm, dmic_audio_map, ARRAY_SIZE(dmic_audio_map)); if (ret) return ret; snd_soc_dapm_enable_pin(dapm, "Digital Mic Legacy"); ret = snd_soc_dapm_sync(dapm); return ret; } static int sdp4430_twl6040_fe_init(struct snd_soc_pcm_runtime *rtd) { /* don't wait before switching of FE power */ rtd->pmdown_time = 0; return 0; } static int sdp4430_bt_init(struct snd_soc_pcm_runtime *rtd) { /* don't wait before switching of BT power */ rtd->pmdown_time = 0; return 0; } static int sdp4430_stream_event(struct snd_soc_dapm_context *dapm) { /* * set DL1 gains dynamically according to the active output * (Headset, Earpiece) and HSDAC power mode */ return sdp4430_set_pdm_dl1_gains(dapm); } /* TODO: make this a separate BT CODEC driver or DUMMY */ static struct snd_soc_dai_driver dai[] = { { .name = "Bluetooth", .playback = { .stream_name = "BT Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "BT Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, /* TODO: make this a separate FM CODEC driver or DUMMY */ { .name = "FM Digital", .playback = { .stream_name = "FM Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "FM Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { .name = "HDMI", .playback = { .stream_name = "HDMI Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, }, }, }; struct snd_soc_dsp_link fe_media = { .playback = true, .capture = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; struct snd_soc_dsp_link fe_media_capture = { .capture = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; struct snd_soc_dsp_link fe_tones = { .playback = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; struct snd_soc_dsp_link fe_vib = { .playback = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; struct snd_soc_dsp_link fe_modem = { .playback = true, .capture = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; struct snd_soc_dsp_link fe_lp_media = { .playback = true, .trigger = {SND_SOC_DSP_TRIGGER_BESPOKE, SND_SOC_DSP_TRIGGER_BESPOKE}, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp4430_dai[] = { /* * Frontend DAIs - i.e. userspace visible interfaces (ALSA PCMs) */ { .name = "SDP4430 Media", .stream_name = "Multimedia", /* ABE components - MM-UL & MM_DL */ .cpu_dai_name = "MultiMedia1", .platform_name = "omap-pcm-audio", .dynamic = 1, /* BE is dynamic */ .init = sdp4430_twl6040_fe_init, .dsp_link = &fe_media, }, { .name = "SDP4430 Media Capture", .stream_name = "Multimedia Capture", /* ABE components - MM-UL2 */ .cpu_dai_name = "MultiMedia2", .platform_name = "omap-pcm-audio", .dynamic = 1, /* BE is dynamic */ .dsp_link = &fe_media_capture, }, { .name = "SDP4430 Voice", .stream_name = "Voice", /* ABE components - VX-UL & VX-DL */ .cpu_dai_name = "Voice", .platform_name = "omap-pcm-audio", .dynamic = 1, /* BE is dynamic */ .dsp_link = &fe_media, .no_host_mode = SND_SOC_DAI_LINK_OPT_HOST, }, { .name = "SDP4430 Tones Playback", .stream_name = "Tone Playback", /* ABE components - TONES_DL */ .cpu_dai_name = "Tones", .platform_name = "omap-pcm-audio", .dynamic = 1, /* BE is dynamic */ .dsp_link = &fe_tones, }, { .name = "SDP4430 Vibra Playback", .stream_name = "VIB-DL", /* ABE components - DMIC UL 2 */ .cpu_dai_name = "Vibra", .platform_name = "omap-pcm-audio", .dynamic = 1, /* BE is dynamic */ .dsp_link = &fe_vib, }, { .name = "SDP4430 MODEM", .stream_name = "MODEM", /* ABE components - MODEM <-> McBSP2 */ .cpu_dai_name = "MODEM", .platform_name = "aess", .dynamic = 1, /* BE is dynamic */ .init = sdp4430_twl6040_fe_init, .dsp_link = &fe_modem, .ops = &sdp4430_modem_ops, .no_host_mode = SND_SOC_DAI_LINK_NO_HOST, .ignore_suspend = 1, }, { .name = "SDP4430 Media LP", .stream_name = "Multimedia", /* ABE components - MM-DL (mmap) */ .cpu_dai_name = "MultiMedia1 LP", .platform_name = "aess", .dynamic = 1, /* BE is dynamic */ .dsp_link = &fe_lp_media, }, { .name = "Legacy McBSP", .stream_name = "Multimedia", /* ABE components - MCBSP2 - MM-EXT */ .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "omap-pcm-audio", /* FM */ .codec_dai_name = "FM Digital", .no_codec = 1, /* TODO: have a dummy CODEC */ .ops = &sdp4430_mcbsp_ops, .ignore_suspend = 1, }, { .name = "Legacy McPDM", .stream_name = "Headset Playback", /* ABE components - DL1 */ .cpu_dai_name = "mcpdm-dl", .platform_name = "omap-pcm-audio", /* Phoenix - DL1 DAC */ .codec_dai_name = "twl6040-dl1", .codec_name = "twl6040-codec", .ops = &sdp4430_mcpdm_ops, .ignore_suspend = 1, }, { .name = "Legacy DMIC", .stream_name = "DMIC Capture", /* ABE components - DMIC0 */ .cpu_dai_name = "omap-dmic-dai-0", .platform_name = "omap-pcm-audio", /* DMIC codec */ .codec_dai_name = "dmic-hifi", .codec_name = "dmic-codec.0", .init = sdp4430_dmic_init, .ops = &sdp4430_dmic_ops, .ignore_suspend = 1, }, /* * Backend DAIs - i.e. dynamically matched interfaces, invisible to userspace. * Matched to above interfaces at runtime, based upon use case. */ { .name = OMAP_ABE_BE_PDM_DL1, .stream_name = "HS Playback", /* ABE components - DL1 */ .cpu_dai_name = "mcpdm-dl1", .platform_name = "aess", /* Phoenix - DL1 DAC */ .codec_dai_name = "twl6040-dl1", .codec_name = "twl6040-codec", .no_pcm = 1, /* don't create ALSA pcm for this */ .init = sdp4430_twl6040_init, .ops = &sdp4430_mcpdm_ops, .be_id = OMAP_ABE_DAI_PDM_DL1, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_PDM_UL1, .stream_name = "Analog Capture", /* ABE components - UL1 */ .cpu_dai_name = "mcpdm-ul1", .platform_name = "aess", /* Phoenix - UL ADC */ .codec_dai_name = "twl6040-ul", .codec_name = "twl6040-codec", .no_pcm = 1, /* don't create ALSA pcm for this */ .ops = &sdp4430_mcpdm_ops, .be_id = OMAP_ABE_DAI_PDM_UL, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_PDM_DL2, .stream_name = "HF Playback", /* ABE components - DL2 */ .cpu_dai_name = "mcpdm-dl2", .platform_name = "aess", /* Phoenix - DL2 DAC */ .codec_dai_name = "twl6040-dl2", .codec_name = "twl6040-codec", .no_pcm = 1, /* don't create ALSA pcm for this */ .init = sdp4430_twl6040_dl2_init, .ops = &sdp4430_mcpdm_ops, .be_id = OMAP_ABE_DAI_PDM_DL2, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_PDM_VIB, .stream_name = "Vibra", /* ABE components - VIB1 DL */ .cpu_dai_name = "mcpdm-vib", .platform_name = "aess", /* Phoenix - PDM to PWM */ .codec_dai_name = "twl6040-vib", .codec_name = "twl6040-codec", .no_pcm = 1, /* don't create ALSA pcm for this */ .ops = &sdp4430_mcpdm_ops, .be_id = OMAP_ABE_DAI_PDM_VIB, }, { .name = OMAP_ABE_BE_BT_VX_UL, .stream_name = "BT Capture", /* ABE components - MCBSP1 - BT-VX */ .cpu_dai_name = "omap-mcbsp-dai.0", .platform_name = "aess", /* Bluetooth */ .codec_dai_name = "Bluetooth", .no_pcm = 1, /* don't create ALSA pcm for this */ .no_codec = 1, /* TODO: have a dummy CODEC */ .be_hw_params_fixup = mcbsp_be_hw_params_fixup, .ops = &sdp4430_mcbsp_ops, .be_id = OMAP_ABE_DAI_BT_VX, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_BT_VX_DL, .stream_name = "BT Playback", /* ABE components - MCBSP1 - BT-VX */ .cpu_dai_name = "omap-mcbsp-dai.0", .platform_name = "aess", /* Bluetooth */ .codec_dai_name = "Bluetooth", .no_pcm = 1, /* don't create ALSA pcm for this */ .no_codec = 1, /* TODO: have a dummy CODEC */ .init = sdp4430_bt_init, .be_hw_params_fixup = mcbsp_be_hw_params_fixup, .ops = &sdp4430_mcbsp_ops, .be_id = OMAP_ABE_DAI_BT_VX, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_MM_EXT0, .stream_name = "FM Playback", /* ABE components - MCBSP2 - MM-EXT */ .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "aess", /* FM */ .codec_dai_name = "FM Digital", .no_pcm = 1, /* don't create ALSA pcm for this */ .no_codec = 1, /* TODO: have a dummy CODEC */ .be_hw_params_fixup = mcbsp_be_hw_params_fixup, .ops = &sdp4430_mcbsp_ops, .be_id = OMAP_ABE_DAI_MM_FM, }, { .name = OMAP_ABE_BE_MM_EXT1, .stream_name = "MODEM", /* ABE components - MCBSP2 - MM-EXT */ .cpu_dai_name = "omap-mcbsp-dai.1", .platform_name = "aess", /* MODEM */ .codec_dai_name = "MODEM", .no_pcm = 1, /* don't create ALSA pcm for this */ .no_codec = 1, /* TODO: have a dummy CODEC */ .be_hw_params_fixup = mcbsp_be_hw_params_fixup, .ops = &sdp4430_mcbsp_ops, .be_id = OMAP_ABE_DAI_MODEM, .ignore_suspend = 1, }, { .name = OMAP_ABE_BE_DMIC0, .stream_name = "DMIC0 Capture", /* ABE components - DMIC UL 1 */ .cpu_dai_name = "omap-dmic-abe-dai-0", .platform_name = "aess", /* DMIC 0 */ .codec_dai_name = "dmic-hifi", .codec_name = "dmic-codec.0", .ops = &sdp4430_dmic_ops, .no_pcm = 1, /* don't create ALSA pcm for this */ .be_hw_params_fixup = dmic_be_hw_params_fixup, .be_id = OMAP_ABE_DAI_DMIC0, }, { .name = OMAP_ABE_BE_DMIC1, .stream_name = "DMIC1 Capture", /* ABE components - DMIC UL 1 */ .cpu_dai_name = "omap-dmic-abe-dai-1", .platform_name = "aess", /* DMIC 1 */ .codec_dai_name = "dmic-hifi", .codec_name = "dmic-codec.1", .ops = &sdp4430_dmic_ops, .no_pcm = 1, /* don't create ALSA pcm for this */ .be_hw_params_fixup = dmic_be_hw_params_fixup, .be_id = OMAP_ABE_DAI_DMIC1, }, { .name = OMAP_ABE_BE_DMIC2, .stream_name = "DMIC2 Capture", /* ABE components - DMIC UL 2 */ .cpu_dai_name = "omap-dmic-abe-dai-2", .platform_name = "aess", /* DMIC 2 */ .codec_dai_name = "dmic-hifi", .codec_name = "dmic-codec.2", .ops = &sdp4430_dmic_ops, .no_pcm = 1, /* don't create ALSA pcm for this */ .be_hw_params_fixup = dmic_be_hw_params_fixup, .be_id = OMAP_ABE_DAI_DMIC2, }, { .name = OMAP_ABE_BE_VXREC, .stream_name = "VXREC Capture", /* ABE components - VxREC */ .cpu_dai_name = "omap-abe-vxrec-dai", .platform_name = "aess", /* no codec needed */ .codec_dai_name = "null-codec-dai", .no_pcm = 1, /* don't create ALSA pcm for this */ .no_codec = 1, .be_id = OMAP_ABE_DAI_VXREC, .ignore_suspend = 1, }, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .driver_name = "OMAP4", .long_name = "TI OMAP4 Board", .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), .stream_event = sdp4430_stream_event, }; static struct platform_device *sdp4430_snd_device; static int __init sdp4430_soc_init(void) { int ret; if (!machine_is_omap_4430sdp() && !machine_is_omap4_panda() && !machine_is_omap_tabletblaze()) { pr_debug("Not SDP4430, BlazeTablet or PandaBoard!\n"); return -ENODEV; } printk(KERN_INFO "SDP4430 SoC init\n"); if (machine_is_omap_4430sdp()) snd_soc_sdp4430.name = "SDP4430"; else if (machine_is_omap4_panda()) snd_soc_sdp4430.name = "Panda"; else if (machine_is_omap_tabletblaze()) snd_soc_sdp4430.name = "Tablet44xx"; sdp4430_snd_device = platform_device_alloc("soc-audio", -1); if (!sdp4430_snd_device) { printk(KERN_ERR "Platform device allocation failed\n"); return -ENOMEM; } ret = snd_soc_register_dais(&sdp4430_snd_device->dev, dai, ARRAY_SIZE(dai)); if (ret < 0) goto err; platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); ret = platform_device_add(sdp4430_snd_device); if (ret) goto err_dev; twl6040_codec = snd_soc_card_get_codec(&snd_soc_sdp4430, "twl6040-codec"); if(twl6040_codec <= 0) { printk(KERN_ERR "sdp4430: could not find `twl6040-codec`\n"); ret = -ENODEV; goto err_dev; } av_switch_reg = regulator_get(&sdp4430_snd_device->dev, "av-switch"); if (IS_ERR(av_switch_reg)) { ret = PTR_ERR(av_switch_reg); printk(KERN_ERR "couldn't get AV Switch regulator %d\n", ret); goto err_dev; } /* Default mode is low-power, MCLK not required */ twl6040_power_mode = 0; cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); /* * CDC CLK2 supplies TWL6040 MCLK, drive it from REQ2INT to * have full control of MCLK gating */ cdc_tcxo_set_req_prio(CDC_TCXO_CLK2, CDC_TCXO_PRIO_REQINT); return ret; err_dev: snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai)); err: platform_device_put(sdp4430_snd_device); return ret; } module_init(sdp4430_soc_init); static void __exit sdp4430_soc_exit(void) { regulator_put(av_switch_reg); cdc_tcxo_set_req_int(CDC_TCXO_CLK2, 0); cdc_tcxo_set_req_prio(CDC_TCXO_CLK2, CDC_TCXO_PRIO_REQINT); platform_device_unregister(sdp4430_snd_device); snd_soc_unregister_dais(&sdp4430_snd_device->dev, ARRAY_SIZE(dai)); } module_exit(sdp4430_soc_exit); MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("ALSA SoC SDP4430"); MODULE_LICENSE("GPL");