MP3 Player. You can change fwd/rev speed and skip. see: http://mbed.org/users/okini3939/notebook/lpc4088_madplayer/
Dependencies: I2SSlave SDFileSystem TLV320 mbed
Diff: player.cpp
- Revision:
- 0:8ba6230eefbd
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/player.cpp Tue Feb 18 00:22:50 2014 +0000 @@ -0,0 +1,314 @@ +#include "mbed.h" +#include "player.h" +#include "SDFileSystem.h" +#include "sdram.h" + +#define DACBUF_SIZE (1024 * 1024 * 8 - 1024) +SDFileSystem sd(p5, p6, p7, p8, "sd"); +TLV320 audio(p32, p31, 0x34, p11, p12, p13, p14, p16); // I2S Codec / sda, scl, addr, tx_sda, tx_ws, clk, rx_sda, rx_ws + +static struct mad_decoder decoder; +struct dacout_s *dacbuf; +volatile int dac_r, dac_w, dac_l; +FILE *fp; +volatile int player_busy = 0, cmd_stop = 0; +int dac_step = 1, dac_vol = 100; + +extern DigitalOut led1, led2, led3, led4; +extern Serial pc; + + +bool isFull() { + return (((dac_w + 1) % DACBUF_SIZE) == dac_r); +}; +bool isEmpty() { + return (dac_r == dac_w); +}; +bool isEmpty2() { + return (dac_r == dac_l); +}; +uint32_t available() { + return (dac_w >= dac_r) ? dac_w - dac_r : DACBUF_SIZE - dac_r + dac_w; +}; +uint32_t available2() { + return (dac_r >= dac_l) ? dac_r - dac_l : DACBUF_SIZE - dac_l + dac_r; +}; + + +void isr_audio () +{ + int i, j, a; + static int buf[4] = {0,0,0,0}; + static int w = 0; + static short l = 0, r = 0; + + for (i = 0; i < 4; i ++) { + if (dac_step > 0 && !isEmpty()) { + // fwd + for (j = 0; j < dac_step; j ++) { +// buf[i] = (dacbuf[dac_r].l << 16) | dacbuf[dac_r].r; + l = dacbuf[dac_r].l; + r = dacbuf[dac_r].r; + buf[i] = (l << 16) | (r & 0xffff); + dac_r = (dac_r + 1) % DACBUF_SIZE; + if (isEmpty()) break; + } + } else + if (dac_step < 0 && !isEmpty2()) { + // rev + for (j = 0; j < -dac_step; j ++) { +// buf[i] = (dacbuf[dac_r].l << 16) | dacbuf[dac_r].r; + l = dacbuf[dac_r].l; + r = dacbuf[dac_r].r; + buf[i] = (l << 16) | (r & 0xffff); + dac_r = (dac_r - 1 + DACBUF_SIZE) % DACBUF_SIZE; + if (isEmpty2()) break; + } + } else { + // under flow + if (l > 0) l --; + if (l < 0) l ++; + if (r > 0) r --; + if (r < 0) r ++; + buf[i] = (l << 16) | (r & 0xffff); + } + } + audio.write(buf, 0, 4); +} + +int init_audio () { + if (sdram_init() == 1) { + pc.printf("Failed to initialize SDRAM\n"); + return -1; + } + malloc(16); + dacbuf = (dacout_s*)malloc(sizeof(dacout_s) * DACBUF_SIZE); + if (dacbuf == NULL) return -1; + pc.printf("memory %08x\r\n", dacbuf); + + audio.power(0x02); // mic off + audio.inputVolume(0, 0); + audio.frequency(44100); + audio.attach(&isr_audio); + audio.start(TRANSMIT); + NVIC_SetPriority(I2S_IRQn, 1); + NVIC_SetPriority(TIMER3_IRQn, 10); + return 0; +} + +int play (char *filename) { + int i; + + DBG("play: %s\r\n", filename); + fp = fopen(filename, "rb"); + if(!fp) { + pc.printf("file error\r\n"); + return -1; + } + player_busy = 1; + dac_r = dac_w = dac_l = 0; + dac_step = 1; + cmd_stop = 0; + led2 = 0; + mad_decoder_init(&decoder, NULL, input, 0, 0, output, error_fn, 0); + mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC); + mad_decoder_finish(&decoder); + fclose(fp); + fp = 0; + dac_r = dac_w = 0; + led2 = 1; + player_busy = 0; + wait_ms(100); + DBG("eof\r\n"); + return 0; +} + +int command (char *cmd) { + int i, r = -1; + + pc.printf("command %s\r\n", cmd); + switch (cmd[0]) { + case 'P': + if (!player_busy) { + char buf[40]; + strcpy(buf, "/sd/"); + strcat(buf, &cmd[1]); + r = play(buf); + } + break; + case 'S': + cmd_stop = 1; + r = 0; + break; + case 'T': + i = atoi(&cmd[1]); + if (i < -10 || i > 10) break; + dac_step = i; + DBG("dac_step %d\r\n", dac_step); + r = 0; + break; + case 'Q': + if (cmd[1] == '+' || cmd[1] == '-') { + i = atof(&cmd[1]) * 44100; + if (i < 0 && i < - available2()) break; + if (i > 0 && i > available()) break; + dac_r += i; + DBG("skip %+d\r\n", i); + } else { + i = atof(&cmd[1]) * 44100; + if (i < dac_l || i > dac_r + available()) break; + dac_r = i; + DBG("skip %d\r\n", i); + } + r = 0; + break; + case 'V': + i = atoi(&cmd[1]); + if (i < 0 || i > 200) break; + dac_vol = i; + DBG("volume %d\r\n", i); + r = 0; + break; + } + + return r; +} + + +/* + * This is the input callback. The purpose of this callback is to (re)fill + * the stream buffer which is to be decoded. + */ + +enum mad_flow input(void *data, + struct mad_stream *stream) +{ + static unsigned char strmbuff[2100]; + int ret; + int rsz; + unsigned char *bp; + + /* the remaining bytes from incomplete frames must be copied + to the beginning of the new buffer ! + */ + bp = strmbuff; + rsz = 0; + if(stream->error == MAD_ERROR_BUFLEN||stream->buffer==NULL) { + if(stream->next_frame!=NULL) { + rsz = stream->bufend-stream->next_frame; + memmove(strmbuff,stream->next_frame,rsz); + bp = strmbuff+rsz; + } + } + + if (feof(fp)) { + if (isEmpty()) { + return MAD_FLOW_STOP; + } else { + return MAD_FLOW_CONTINUE; + } + } + + led4 = 1; + ret = fread(bp,1,sizeof(strmbuff) - rsz,fp); + + if (!ret) { + DBG("input stop\r\n"); + return MAD_FLOW_STOP; + } + + mad_stream_buffer(stream, strmbuff, ret + rsz); + + return MAD_FLOW_CONTINUE; +} + + +/* + * The following utility routine performs simple rounding, clipping, and + * scaling of MAD's high-resolution samples down to 16 bits. It does not + * perform any dithering or noise shaping, which would be recommended to + * obtain any exceptional audio quality. It is therefore not recommended to + * use this routine if high-quality output is desired. + */ + +static /*inline*/ +signed int scale(mad_fixed_t sample) +{ + /* round */ + sample += (1L << (MAD_F_FRACBITS - 16)); + + /* clip */ + if (sample >= MAD_F_ONE) + sample = MAD_F_ONE - 1; + else if (sample < -MAD_F_ONE) + sample = -MAD_F_ONE; + + /* quantize */ + return sample >> (MAD_F_FRACBITS + 1 - 16); +} + +/* + * This is the output callback function. It is called after each frame of + * MPEG audio data has been completely decoded. The purpose of this callback + * is to output (or play) the decoded PCM audio. + */ + +enum mad_flow output(void *data, + struct mad_header const *header, + struct mad_pcm *pcm) +{ + unsigned int nchannels, nsamples; + mad_fixed_t const *left_ch, *right_ch; + + /* pcm->samplerate contains the sampling frequency */ + nchannels = pcm->channels; + nsamples = pcm->length; + left_ch = pcm->samples[0]; + right_ch = pcm->samples[1]; + + poll(); + while (nsamples--) { + while (isFull() || available() >= (44100 * FWDBUF)) { + poll(); + if (cmd_stop) break; + } + __disable_irq(); + dacbuf[dac_w].l = scale(*left_ch); + dacbuf[dac_w].r = scale(*right_ch); + dac_w = (dac_w + 1) % DACBUF_SIZE; + if (dac_w == 0 || dac_l) dac_l ++; + __enable_irq(); + left_ch++; + right_ch++; + } + + if (cmd_stop) { + DBG("output stop o\r\n"); + cmd_stop = 0; + return MAD_FLOW_STOP; + } + return MAD_FLOW_CONTINUE; +} + +/* + * This is the error callback function. It is called whenever a decoding + * error occurs. The error is indicated by stream->error; the list of + * possible MAD_ERROR_* errors can be found in the mad.h (or stream.h) + * header file. + */ + +enum mad_flow error_fn(void *data, + struct mad_stream *stream, + struct mad_frame *frame) +{ + /* ID3 tags will cause warnings and short noise, ignore it for the moment*/ + /* + fprintf(stderr, "decoding error 0x%04x (%s)\n", + stream->error, mad_stream_errorstr(stream)); + */ + + /* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */ + + return MAD_FLOW_CONTINUE; +} +