【MM32F5270开发板试用】六、如何用 星辰内核 + 国产RTOS 通过I2S播放 “星辰大海”

本篇文章来自极术社区与灵动组织的MM32F5270开发板评测活动，更多开发板试用活动请关注极术社区网站。作者：Magicoe是攻城狮

这个demo和想法是参考了大神的文章
https://aijishu.com/a/1060000…

我简单的想法是 和 星辰 这个内核呼应下，星辰大海嘛，这首歌还挺好听的，咱们的目标是星辰大海，目标要大一点 应该能实现。

先说音频播放的接口，一般是I2S，这一点大神的文章里介绍的很详细了，我就不再赘述。我关注的目标是rt-thread上播放音乐这码事，稍微区别下。

rt-thread提供了wav播放的包，名字叫wavplayer，我懒得移植helix的MP3
库了，估计有很多人移植来着，我就不趟了，我的核心是IoT方向，今早收拢战线回归主要战场。
wavplayer的下载链接，当然如果你有空用脚本生成那倒无所谓
https://github.com/RT-Thread-…
或
gitee我没找到…
在keil的工程里添加wavplayer的源码
这里我没有用到record录音的功能，所以这部分c文件就编译屏蔽了

wavplayer需要用到rtt的optparse的相关定义和函数，添加optparse.c及其include的路径到工程

接下来就是rt-thread的audio框架的相关代码了，在component文件夹下
主要是audio.c和audio_pipe.c

嗯，预想实现功能，rtconfig.h的修改是必不可少的，咱们需要打开如下几个宏定义，欸~

一切就绪就是本次文章的重头了，也是我干到夜里2点没干动的代码—drv_sound.c
我先把源文件列在这里，再讲我的思路，整的不好容易 啪啪啪 的破音，破音要么是数据DMA搬运的不连续，要么就是数据有问题，反正多搜索网上的帖子就好。
我代码偷懒了，并没有实现audio频率的设置，音量的设置（板子不支持）等我觉得很烦的功能，懒的搞，能播就行

/* * Copyright (c) 2020-2021, Bluetrum Development Team * * SPDX-License-Identifier: Apache-2.0 * * Date    AuthorNotes * 2020-12-12     greedyhao    first implementation */#include #include "rtdevice.h"#define DBG_TAG"drv.snd_dev"#define DBG_LVLDBG_ERROR#include #include #include #include "hal_common.h"#include "hal_rcc.h"#include "hal_i2s.h"#include "hal_dma.h"#include "hal_dma_request.h"#include "hal_gpio.h"#include "clock_init.h"#define SAI_AUDIO_FREQUENCY_48K  ((uint32_t)48000u)#define SAI_AUDIO_FREQUENCY_44K  ((uint32_t)44100u)#define SAI_AUDIO_FREQUENCY_38K  ((uint32_t)38000u)#define TX_FIFO_SIZE      (4096*2)struct sound_device{    struct rt_audio_device audio;    struct rt_audio_configure replay_config;    rt_uint8_t *tx_fifo;    rt_uint8_t  volume;};static struct sound_device snd_dev = {0};#pragma pack (4)volatile uint8_t g_PlayIndex = 0;uint8_t g_AudioBuf[TX_FIFO_SIZE] __attribute__((section(".ARM.__at_0x20000000"))) ;#pragma pack()/* I2S IRQ. */void DMA1_CH5_IRQHandler(void){    rt_interrupt_enter(); if (0u != (DMA_GetChannelInterruptStatus(DMA1, DMA_REQ_DMA1_SPI2_TX) & DMA_CHN_INT_XFER_DONE) )    { DMA_ClearChannelInterruptStatus(DMA1, DMA_REQ_DMA1_SPI2_TX, DMA_CHN_INT_XFER_DONE); DMA_EnableChannel(DMA1, DMA_REQ_DMA1_SPI2_TX, true);  g_PlayIndex = 1; rt_audio_tx_complete(&snd_dev.audio);     } if(0u != (DMA_CHN_INT_XFER_HALF_DONE & DMA_GetChannelInterruptStatus(DMA1, DMA_REQ_DMA1_SPI2_TX)) )    { DMA_ClearChannelInterruptStatus(DMA1, DMA_REQ_DMA1_SPI2_TX, DMA_CHN_INT_XFER_HALF_DONE); g_PlayIndex = 0; rt_audio_tx_complete(&snd_dev.audio);    } rt_interrupt_leave();}//apll = 采样率*ADPLL_DIV*512//audio pll initvoid adpll_init(uint8_t out_spr){    GPIO_Init_Type gpio_init; /* SPI2. */    RCC_EnableAPB1Periphs(RCC_APB1_PERIPH_SPI2, true);    RCC_ResetAPB1Periphs(RCC_APB1_PERIPH_SPI2); /* PD3 - I2S_CK. */    gpio_init.Pins  = GPIO_PIN_3;    gpio_init.PinMode  = GPIO_PinMode_AF_PushPull;    gpio_init.Speed = GPIO_Speed_10MHz;    GPIO_Init(GPIOD, &gpio_init);    GPIO_PinAFConf(GPIOD, gpio_init.Pins, GPIO_AF_5);    /* PE6 - I2S_SD. */    gpio_init.Pins  = GPIO_PIN_6;    gpio_init.PinMode  = GPIO_PinMode_AF_PushPull;    gpio_init.Speed = GPIO_Speed_10MHz;    GPIO_Init(GPIOE, &gpio_init);    GPIO_PinAFConf(GPIOE, gpio_init.Pins, GPIO_AF_5);    /* PE4 - I2S_WS. */    gpio_init.Pins  = GPIO_PIN_4;    gpio_init.PinMode  = GPIO_PinMode_AF_PushPull;    gpio_init.Speed = GPIO_Speed_10MHz;    GPIO_Init(GPIOE, &gpio_init);    GPIO_PinAFConf(GPIOE, gpio_init.Pins, GPIO_AF_5);    /* PE5 - I2S_MCK. */    gpio_init.Pins  = GPIO_PIN_5;    gpio_init.PinMode  = GPIO_PinMode_AF_PushPull;    gpio_init.Speed = GPIO_Speed_10MHz;    GPIO_Init(GPIOE, &gpio_init);    GPIO_PinAFConf(GPIOE, gpio_init.Pins, GPIO_AF_5);     /* Setup the DMA for I2S RX. */    DMA_Channel_Init_Type dma_channel_init;    dma_channel_init.MemAddr    = (uint32_t)(g_AudioBuf);    dma_channel_init.MemAddrIncMode    = DMA_AddrIncMode_IncAfterXfer;    dma_channel_init.PeriphAddr = I2S_GetTxDataRegAddr(SPI2);  /* use tx data register here. */    dma_channel_init.PeriphAddrIncMode = DMA_AddrIncMode_StayAfterXfer;    dma_channel_init.Priority   = DMA_Priority_Highest;    dma_channel_init.XferCount  = TX_FIFO_SIZE/2;    dma_channel_init.XferMode   = DMA_XferMode_MemoryToPeriph;    dma_channel_init.ReloadMode = DMA_ReloadMode_AutoReload;    dma_channel_init.XferWidth  = DMA_XferWidth_16b;    DMA_InitChannel(DMA1, DMA_REQ_DMA1_SPI2_TX, &dma_channel_init);    /* Enable DMA transfer done interrupt. */    DMA_EnableChannelInterrupts(DMA1, DMA_REQ_DMA1_SPI2_TX, DMA_CHN_INT_XFER_DONE, true);    DMA_EnableChannelInterrupts(DMA1, DMA_REQ_DMA1_SPI2_TX, DMA_CHN_INT_XFER_HALF_DONE, true);    NVIC_EnableIRQ(DMA1_CH5_IRQn);    /* Setup the I2S. */    I2S_Master_Init_Type i2s_master_init;    i2s_master_init.ClockFreqHz  = CLOCK_APB1_FREQ;    i2s_master_init.SampleRate   = SAI_AUDIO_FREQUENCY_44K;    i2s_master_init.DataWidth    = I2S_DataWidth_16b;    i2s_master_init.Protocol     = I2S_Protocol_PHILIPS;    i2s_master_init.EnableMCLK   = true;    i2s_master_init.Polarity     = I2S_Polarity_1;    i2s_master_init.XferMode     = I2S_XferMode_TxOnly;    I2S_InitMaster(SPI2, &i2s_master_init);    I2S_EnableDMA(SPI2, true);    I2S_Enable(SPI2, true);    DMA_EnableChannel(DMA1, DMA_REQ_DMA1_SPI2_TX, true);}static rt_err_t sound_getcaps(struct rt_audio_device *audio, struct rt_audio_caps *caps){    rt_err_t result = RT_EOK;    struct sound_device *snd_dev = RT_NULL;    RT_ASSERT(audio != RT_NULL);    snd_dev = (struct sound_device *)audio->parent.user_data;    LOG_D("%s:main_type: %d, sub_type: %d", __FUNCTION__, caps->main_type, caps->sub_type); switch (caps->main_type)    {    case AUDIO_TYPE_QUERY: /* qurey the types of hw_codec device */    { switch (caps->sub_type) { case AUDIO_TYPE_QUERY:     caps->udata.mask = AUDIO_TYPE_OUTPUT | AUDIO_TYPE_MIXER;     break; default:     result = -RT_ERROR;     break; } break;    }    case AUDIO_TYPE_OUTPUT: /* Provide capabilities of OUTPUT unit */    { switch (caps->sub_type) { case AUDIO_DSP_PARAM:     caps->udata.config.samplerate   = snd_dev->replay_config.samplerate;     caps->udata.config.channels     = snd_dev->replay_config.channels;     caps->udata.config.samplebits   = snd_dev->replay_config.samplebits;     break; case AUDIO_DSP_SAMPLERATE:     caps->udata.config.samplerate   = snd_dev->replay_config.samplerate;     break; case AUDIO_DSP_CHANNELS:     caps->udata.config.channels     = snd_dev->replay_config.channels;     break; case AUDIO_DSP_SAMPLEBITS:     caps->udata.config.samplebits   = snd_dev->replay_config.samplebits;     break; default:     result = -RT_ERROR;     break; } break;    }    case AUDIO_TYPE_MIXER: /* report the Mixer Units */    { switch (caps->sub_type) { case AUDIO_MIXER_QUERY:     caps->udata.mask = AUDIO_MIXER_VOLUME;     break; case AUDIO_MIXER_VOLUME:   //  caps->udata.value =  saia_volume_get();     break; default:     result = -RT_ERROR;     break; } break;    }    default: result = -RT_ERROR; break;    }    return RT_EOK;}static rt_err_t sound_configure(struct rt_audio_device *audio, struct rt_audio_caps *caps){    rt_err_t result = RT_EOK;    struct sound_device *snd_dev = RT_NULL;    RT_ASSERT(audio != RT_NULL);    snd_dev = (struct sound_device *)audio->parent.user_data;    switch (caps->main_type)    {    case AUDIO_TYPE_MIXER:    { switch (caps->sub_type) { case AUDIO_MIXER_VOLUME: {     rt_uint8_t volume = caps->udata.value;  //   saia_volume_set(volume);     snd_dev->volume = volume;     LOG_D("set volume %d", volume);     break; } case AUDIO_MIXER_EXTEND: break; default:     result = -RT_ERROR;     break; } break;    }    case AUDIO_TYPE_OUTPUT:    { switch (caps->sub_type) { case AUDIO_DSP_PARAM: {     /* set samplerate */   //  saia_frequency_set(caps->udata.config.samplerate);     /* set channels */   //  saia_channels_set(caps->udata.config.channels);     /* save configs */     snd_dev->replay_config.samplerate = caps->udata.config.samplerate;     snd_dev->replay_config.channels   = caps->udata.config.channels;     snd_dev->replay_config.samplebits = caps->udata.config.samplebits;     LOG_D("set samplerate %d", snd_dev->replay_config.samplerate);     break; } case AUDIO_DSP_SAMPLERATE: {  //   saia_frequency_set(caps->udata.config.samplerate);     snd_dev->replay_config.samplerate = caps->udata.config.samplerate;     LOG_D("set samplerate %d", snd_dev->replay_config.samplerate);     break; } case AUDIO_DSP_CHANNELS: {   //  saia_channels_set(caps->udata.config.channels);     snd_dev->replay_config.channels   = caps->udata.config.channels;     LOG_D("set channels %d", snd_dev->replay_config.channels);     break; } case AUDIO_DSP_SAMPLEBITS: {     /* not support */     snd_dev->replay_config.samplebits = caps->udata.config.samplebits;     break; } default:     result = -RT_ERROR;     break; } break;    }    default: break;    }    return RT_EOK;}static rt_err_t sound_init(struct rt_audio_device *audio){    struct sound_device *snd_dev = RT_NULL;    RT_ASSERT(audio != RT_NULL);    snd_dev = (struct sound_device *)audio->parent.user_data;    adpll_init(0);    /* set default params */   // saia_frequency_set(snd_dev->replay_config.samplerate);    //saia_channels_set(snd_dev->replay_config.channels);    //saia_volume_set(snd_dev->volume);    return RT_EOK;}static rt_err_t sound_start(struct rt_audio_device *audio, int stream){    struct sound_device *snd_dev = RT_NULL;    RT_ASSERT(audio != RT_NULL);    snd_dev = (struct sound_device *)audio->parent.user_data;    if (stream == AUDIO_STREAM_REPLAY)    { LOG_D("open sound device"); DMA_EnableChannel(DMA1, DMA_REQ_DMA1_SPI2_TX, true);    }    return RT_EOK;}static rt_err_t sound_stop(struct rt_audio_device *audio, int stream){    RT_ASSERT(audio != RT_NULL);    if (stream == AUDIO_STREAM_REPLAY)    { LOG_D("close sound device"); DMA_EnableChannel(DMA1, DMA_REQ_DMA1_SPI2_TX, false);    }    return RT_EOK;}rt_size_t sound_transmit(struct rt_audio_device *audio, const void *writeBuf, void *readBuf, rt_size_t size){    struct sound_device *snd_dev = RT_NULL;    rt_size_t count = 0;//    RT_ASSERT(audio != RT_NULL);//    snd_dev = (struct sound_device *)audio->parent.user_data;//    rt_kprintf("Size %d   %d\r\n", size, g_PlayIndex);    if(g_PlayIndex == 0)    { memcpy(&g_AudioBuf[0], writeBuf, size);    }    else    { memcpy(&g_AudioBuf[TX_FIFO_SIZE/2], writeBuf, size);    } return size;}static void sound_buffer_info(struct rt_audio_device *audio, struct rt_audio_buf_info *info){    struct sound_device *snd_dev = RT_NULL;    RT_ASSERT(audio != RT_NULL);    snd_dev = (struct sound_device *)audio->parent.user_data;    /**     * TX_FIFO     * +----------------+----------------+     * |     block1     |     block2     |     * +----------------+----------------+     *  \  block_size  /     */    info->buffer      = snd_dev->tx_fifo;    info->total_size  = TX_FIFO_SIZE;    info->block_size  = TX_FIFO_SIZE/2;    info->block_count = 2;}static struct rt_audio_ops ops ={    .getcaps     = sound_getcaps,    .configure   = sound_configure,    .init = sound_init,    .start= sound_start,    .stop = sound_stop,    .transmit    = sound_transmit,//NULL,//sound_transmit,    .buffer_info = sound_buffer_info,};static int rt_hw_sound_init(void){    rt_uint8_t *tx_fifo = RT_NULL;    rt_uint8_t *rx_fifo = RT_NULL;    tx_fifo = rt_malloc(TX_FIFO_SIZE);    if(tx_fifo == RT_NULL)    { rt_kprintf("Sound can alloc tx_fifo\r\n"); return -RT_ENOMEM;    } rt_memset(&g_AudioBuf[0], 0x00, TX_FIFO_SIZE); rt_memset(tx_fifo, 0, TX_FIFO_SIZE/2);    snd_dev.tx_fifo = tx_fifo;    /* init default configuration */    { snd_dev.replay_config.samplerate = SAI_AUDIO_FREQUENCY_44K; snd_dev.replay_config.channels   = 2; snd_dev.replay_config.samplebits = 16; snd_dev.volume     = 55;    }    /* register snd_dev device */    snd_dev.audio.ops = &ops;    rt_audio_register(&snd_dev.audio, "sound0", RT_DEVICE_FLAG_WRONLY, &snd_dev);    return RT_EOK;}INIT_DEVICE_EXPORT(rt_hw_sound_init);

DMA搬运wav的音频数据我遇到了很多问题，最后用一个大的buffer给DMA搬运，把这个buffer拆成两半，DMA使用half transfer interrupt以及transfered interrupt即传输一半给中断，传输完成给中断。
传输一半的时候让wav解码任务读取一部分内容放到buffer前半部分，传输完成通知wav解码任务再读取一部分内容放到buffer的后半部分。类似pingpong buffer 这种机制，才能保证wav播放的连续性。

好吧编译后下载运行看视频的结果，wav播放的命令是wavplayer -s 文件名