484 lines
12 KiB
C
484 lines
12 KiB
C
/*
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* Copyright (c) 2020-2021, Bluetrum Development Team
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*
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* SPDX-License-Identifier: Apache-2.0
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*
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* Date Author Notes
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* 2020-12-12 greedyhao first implementation
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*/
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#include <board.h>
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#define DBG_TAG "drv.snd_dev"
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#define DBG_LVL DBG_INFO
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#include <rtdbg.h>
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#define SAI_AUDIO_FREQUENCY_44K ((uint32_t)44100u)
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#define SAI_AUDIO_FREQUENCY_48K ((uint32_t)48000u)
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#define TX_FIFO_SIZE (1024)
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struct sound_device
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{
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struct rt_audio_device audio;
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struct rt_audio_configure replay_config;
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rt_uint8_t *tx_fifo;
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rt_uint8_t *rx_fifo;
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rt_uint8_t volume;
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};
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static struct sound_device snd_dev = {0};
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//apll = 采样率*ADPLL_DIV*512
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//audio pll init
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void adpll_init(uint8_t out_spr)
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{
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PLL1CON &= ~(BIT(16) | BIT(17)); //PLL1 refclk select xosc26m
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CLKCON2 &= ~(BIT(4)| BIT(5) | BIT(6) | BIT(7));
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PLL1CON &= ~(BIT(3) | BIT(4) | BIT(5));
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PLL1CON |= BIT(3); //Select PLL/VCO frequency band (PLL大于206M vcos = 0x01, 否则为0)
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PLL1CON |= BIT(12); //enable pll1 ldo
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hal_mdelay(1);
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PLL1CON |= BIT(18); //pll1 sdm enable
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if (out_spr) {
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CLKCON2 |= BIT(4) | BIT(7); //adpll_div = 10
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PLL1DIV = (245.76 * 65536) / 26; //245.76Mhz for 48K
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// sys.aupll_type = 1;
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} else {
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CLKCON2 |= BIT(5) | BIT(7); //adpll_div = 11
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PLL1DIV = (248.3712 * 65536) / 26; //248.3712MHz for 44.1k
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// sys.aupll_type = 0;
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}
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hal_mdelay(1);
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PLL1CON |= BIT(20); //update pll1div
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PLL1CON |= BIT(6); //enable analog pll1
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hal_mdelay(1); //wait pll1 stable
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}
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void dac_start(void)
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{
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AUANGCON0 |= BIT(0) | BIT(1) | BIT(3); // bg ldoh bias enable
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AUANGCON0 &= ~(BIT(6)|BIT(5)|BIT(4)); // LDOH voltage select：3bit
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AUANGCON0 |= (3<<4); // 2.4/2.5/2.7/2.9/3.1/3.2
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AUANGCON0 |= BIT(2); // LDOL enable
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AUANGCON0 |= BIT(9); //VCM enable
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AUANGCON0 &= ~(BIT(13)|BIT(12)); // VCM voltage select, 2bit
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AUANGCON0 |= (2<<12);
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AUANGCON0 |= BIT(15) | BIT(16) | BIT(17) | BIT(18); // d2a lpf audpa audpa_dly
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AUANGCON0 &= ~BIT(11); //VCM type: 0-->res divider with off-chip cap; 1-->internal VCM
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//AUANGCON0 |= BIT(11);
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AUANGCON0 &= ~BIT(19); // dac type: 0-->SC; 1-->SR
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//AUANGCON0 |= BIT(19);
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AUANGCON0 |= BIT(20); // pa type: 0-->diff; 1-->3.3V single
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AUANGCON3 &= ~(0x7<<4); //BIT[6:4]=PA_GF[2:0]
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AUANGCON3 |= (0<<4);
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AUANGCON3 &= ~(0xf); //BIT[3:0]=PA_GX[3:0]
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AUANGCON3 |= 0;
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AUANGCON3 &= ~(0xF<<8); //BIT[11:8]=PA2_GX[3:0]
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AUANGCON3 |= (0<<8);
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AUANGCON3 &= ~(0x7<<12); //BIT[14:12]=PA2_GF[2:0]
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AUANGCON3 |= (0<<12);
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AUANGCON1 |= BIT(0) | BIT(1); // dac enable: BIT(0)-->right channel; BIT(1)-->left channel
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//AUANGCON1 &= ~BIT(1); //disable left channel
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AUANGCON1 |= BIT(12); // lpf2pa enable
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AUANGCON1 &= ~BIT(29); // vcmbuf enable: 0-->disable
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//AUANGCON1 |= BIT(29);
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//AUANGCON1 |= BIT(30); // mirror enable
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//AUANGCON2 |= BIT(29) | BIT(30); // adc mute
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//AUANGCON1 |= BIT(3); // pa mute
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}
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void saia_frequency_set(uint32_t frequency)
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{
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if (frequency == SAI_AUDIO_FREQUENCY_48K) {
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DACDIGCON0 |= BIT(1);
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DACDIGCON0 &= ~(0xf << 2);
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DACDIGCON0 |= BIT(6);
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} else if (frequency == SAI_AUDIO_FREQUENCY_44K) {
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DACDIGCON0 &= ~BIT(1);
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DACDIGCON0 &= ~(0xf << 2);
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DACDIGCON0 |= BIT(1);
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DACDIGCON0 |= BIT(6);
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}
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}
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void saia_channels_set(uint8_t channels)
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{
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LOG_D("saia_channels_set=%d", channels);
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if (channels == 1) {
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AU0LMIXCOEF = 0x00007FFF;
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AU1LMIXCOEF = 0x00007FFF;
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DACDIGCON0 |= BIT(7);
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DACDIGCON0 |= BIT(8);
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AUANGCON1 &= ~BIT(0);
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} else {
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AUANGCON1 |= BIT(0);
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DACDIGCON0 &= ~BIT(7);
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DACDIGCON0 &= ~BIT(8);
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}
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}
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void saia_volume_set(rt_uint8_t volume)
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{
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if (volume > 100)
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volume = 100;
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uint32_t dvol = volume * 327; // max is 0x7ffff
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LOG_D("dvol=0x%x", dvol);
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DACVOLCON = dvol | (0x02 << 16); // dac fade in
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}
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uint8_t saia_volume_get(void)
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{
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return ((DACVOLCON & 0xffff) / 327);
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}
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static rt_err_t sound_getcaps(struct rt_audio_device *audio, struct rt_audio_caps *caps)
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{
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rt_err_t result = RT_EOK;
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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switch (caps->main_type)
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{
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case AUDIO_TYPE_QUERY: /* qurey the types of hw_codec device */
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{
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switch (caps->sub_type)
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{
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case AUDIO_TYPE_QUERY:
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caps->udata.mask = AUDIO_TYPE_OUTPUT | AUDIO_TYPE_MIXER;
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break;
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default:
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result = -RT_ERROR;
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break;
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}
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break;
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}
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case AUDIO_TYPE_OUTPUT: /* Provide capabilities of OUTPUT unit */
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{
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switch (caps->sub_type)
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{
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case AUDIO_DSP_PARAM:
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caps->udata.config.samplerate = snd_dev->replay_config.samplerate;
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caps->udata.config.channels = snd_dev->replay_config.channels;
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caps->udata.config.samplebits = snd_dev->replay_config.samplebits;
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break;
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case AUDIO_DSP_SAMPLERATE:
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caps->udata.config.samplerate = snd_dev->replay_config.samplerate;
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break;
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case AUDIO_DSP_CHANNELS:
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caps->udata.config.channels = snd_dev->replay_config.channels;
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break;
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case AUDIO_DSP_SAMPLEBITS:
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caps->udata.config.samplebits = snd_dev->replay_config.samplebits;
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break;
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default:
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result = -RT_ERROR;
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break;
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}
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break;
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}
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case AUDIO_TYPE_MIXER: /* report the Mixer Units */
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{
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switch (caps->sub_type)
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{
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case AUDIO_MIXER_QUERY:
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caps->udata.mask = AUDIO_MIXER_VOLUME;
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break;
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case AUDIO_MIXER_VOLUME:
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caps->udata.value = saia_volume_get();
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break;
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default:
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result = -RT_ERROR;
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break;
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}
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break;
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}
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default:
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result = -RT_ERROR;
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break;
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}
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return RT_EOK;
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}
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static rt_err_t sound_configure(struct rt_audio_device *audio, struct rt_audio_caps *caps)
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{
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rt_err_t result = RT_EOK;
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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switch (caps->main_type)
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{
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case AUDIO_TYPE_MIXER:
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{
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switch (caps->sub_type)
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{
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case AUDIO_MIXER_VOLUME:
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{
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rt_uint8_t volume = caps->udata.value;
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saia_volume_set(volume);
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snd_dev->volume = volume;
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LOG_D("set volume %d", volume);
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break;
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}
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default:
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result = -RT_ERROR;
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break;
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}
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break;
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}
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case AUDIO_TYPE_OUTPUT:
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{
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switch (caps->sub_type)
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{
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case AUDIO_DSP_PARAM:
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{
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/* set samplerate */
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saia_frequency_set(caps->udata.config.samplerate);
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/* set channels */
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saia_channels_set(caps->udata.config.channels);
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/* save configs */
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snd_dev->replay_config.samplerate = caps->udata.config.samplerate;
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snd_dev->replay_config.channels = caps->udata.config.channels;
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snd_dev->replay_config.samplebits = caps->udata.config.samplebits;
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LOG_D("set samplerate %d", snd_dev->replay_config.samplerate);
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break;
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}
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case AUDIO_DSP_SAMPLERATE:
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{
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saia_frequency_set(caps->udata.config.samplerate);
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snd_dev->replay_config.samplerate = caps->udata.config.samplerate;
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LOG_D("set samplerate %d", snd_dev->replay_config.samplerate);
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break;
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}
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case AUDIO_DSP_CHANNELS:
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{
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saia_channels_set(caps->udata.config.channels);
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snd_dev->replay_config.channels = caps->udata.config.channels;
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LOG_D("set channels %d", snd_dev->replay_config.channels);
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break;
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}
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case AUDIO_DSP_SAMPLEBITS:
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{
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/* not support */
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snd_dev->replay_config.samplebits = caps->udata.config.samplebits;
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break;
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}
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default:
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result = -RT_ERROR;
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break;
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}
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break;
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}
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default:
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break;
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}
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return RT_EOK;
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}
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static rt_err_t sound_init(struct rt_audio_device *audio)
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{
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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adpll_init(0);
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dac_start();
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/* set default params */
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saia_frequency_set(snd_dev->replay_config.samplerate);
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saia_channels_set(snd_dev->replay_config.channels);
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return RT_EOK;
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}
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static rt_err_t sound_start(struct rt_audio_device *audio, int stream)
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{
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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if (stream == AUDIO_STREAM_REPLAY)
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{
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LOG_D("open sound device");
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AUBUFSIZE = (TX_FIFO_SIZE / 4 - 1);
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AUBUFSIZE |= (TX_FIFO_SIZE / 8) << 16;
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AUBUFSTARTADDR = DMA_ADR(snd_dev->rx_fifo);
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DACDIGCON0 = BIT(0) | BIT(10); // (0x01<<2)
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DACVOLCON = 0x7fff; // -60DB
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DACVOLCON |= BIT(20);
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AUBUFCON |= BIT(1) | BIT(4);
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}
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return RT_EOK;
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}
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static rt_err_t sound_stop(struct rt_audio_device *audio, int stream)
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{
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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if (stream == AUDIO_STREAM_REPLAY)
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{
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AUBUFCON &= ~BIT(4);
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LOG_D("close sound device");
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}
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return RT_EOK;
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}
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rt_size_t sound_transmit(struct rt_audio_device *audio, const void *writeBuf, void *readBuf, rt_size_t size)
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{
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struct sound_device *snd_dev = RT_NULL;
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rt_size_t tmp_size = size / 4;
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rt_size_t count = 0;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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while (tmp_size-- > 0) {
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while(AUBUFCON & BIT(8)); // aubuf full
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AUBUFDATA = ((const uint32_t *)writeBuf)[count++];
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}
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return size;
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}
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static void sound_buffer_info(struct rt_audio_device *audio, struct rt_audio_buf_info *info)
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{
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struct sound_device *snd_dev = RT_NULL;
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RT_ASSERT(audio != RT_NULL);
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snd_dev = (struct sound_device *)audio->parent.user_data;
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/**
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* TX_FIFO
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* +----------------+----------------+
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* | block1 | block2 |
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* +----------------+----------------+
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* \ block_size /
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*/
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info->buffer = snd_dev->tx_fifo;
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info->total_size = TX_FIFO_SIZE;
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info->block_size = TX_FIFO_SIZE / 2;
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info->block_count = 2;
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}
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static struct rt_audio_ops ops =
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{
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.getcaps = sound_getcaps,
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.configure = sound_configure,
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.init = sound_init,
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.start = sound_start,
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.stop = sound_stop,
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.transmit = sound_transmit,
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.buffer_info = sound_buffer_info,
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};
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void audio_isr(int vector, void *param)
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{
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rt_interrupt_enter();
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//Audio buffer pend
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if (AUBUFCON & BIT(5)) {
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AUBUFCON |= BIT(1); //Audio Buffer Pend Clear
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rt_audio_tx_complete(&snd_dev.audio);
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}
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rt_interrupt_leave();
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}
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static int rt_hw_sound_init(void)
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{
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rt_uint8_t *tx_fifo = RT_NULL;
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rt_uint8_t *rx_fifo = RT_NULL;
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/* 分配 DMA 搬运 buffer */
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tx_fifo = rt_calloc(1, TX_FIFO_SIZE);
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if(tx_fifo == RT_NULL)
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{
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return -RT_ENOMEM;
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}
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snd_dev.tx_fifo = tx_fifo;
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/* 分配 DMA 搬运 buffer */
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rx_fifo = rt_calloc(1, TX_FIFO_SIZE);
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if(rx_fifo == RT_NULL)
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{
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return -RT_ENOMEM;
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}
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snd_dev.rx_fifo = rx_fifo;
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/* init default configuration */
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{
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snd_dev.replay_config.samplerate = 48000;
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snd_dev.replay_config.channels = 2;
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snd_dev.replay_config.samplebits = 16;
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snd_dev.volume = 55;
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}
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/* register snd_dev device */
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snd_dev.audio.ops = &ops;
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rt_audio_register(&snd_dev.audio, "sound0", RT_DEVICE_FLAG_WRONLY, &snd_dev);
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rt_hw_interrupt_install(IRQ_AUBUF0_1_VECTOR, audio_isr, RT_NULL, "au_isr");
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return RT_EOK;
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}
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INIT_DEVICE_EXPORT(rt_hw_sound_init);
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