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This commit is contained in:
Ross Owen
2012-08-30 14:21:17 +01:00
parent e449050b40
commit 99cdee6e2b
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module_usb_audio/audio.xc Executable file
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/**
* @file audio.xc
* @brief XMOS L1/L2 USB 2,0 Audio Reference Design. Audio Functions.
* @author Ross Owen, XMOS Semiconductor Ltd
*
* This thread handles I2S and pars an additional SPDIF Tx thread. It forwards samples to the SPDIF Tx thread.
* Additionally this thread handles clocking and CODEC/DAC/ADC config.
**/
#include <syscall.h>
#include <platform.h>
#include <xs1.h>
#include <xclib.h>
#include <print.h>
#include <xs1_su.h>
#include "clocking.h"
#include "audioports.h"
#include "codec.h"
#include "devicedefines.h"
#include "SpdifTransmit.h"
extern out port p_test;
unsigned g_adcVal = 0;
//#define RAMP_CHECK 1
//#pragma xta command "analyse path i2s_output_l i2s_output_r"
//#pragma xta command "set required - 2000 ns"
//#pragma xta command "analyse path i2s_output_r i2s_output_l"
//#pragma xta command "set required - 2000 ns"
/* I2S Data I/O*/
#if (I2S_CHANS_DAC != 0)
extern buffered out port:32 p_i2s_dac[I2S_WIRES_DAC];
#endif
#if (I2S_CHANS_ADC != 0)
extern buffered in port:32 p_i2s_adc[I2S_WIRES_ADC];
#endif
/* I2S LR/Bit clock I/O */
#ifdef CODEC_SLAVE
extern buffered out port:32 p_lrclk;
extern buffered out port:32 p_bclk;
#else
extern in port p_lrclk;
extern in port p_bclk;
#endif
/* Master clock input */
extern port p_mclk;
#ifdef SPDIF
extern buffered out port:32 p_spdif_tx;
#endif
extern clock clk_audio_mclk;
extern clock clk_audio_bclk;
extern clock clk_mst_spd;
extern void device_reboot(void);
/* I2S delivery thread */
#pragma unsafe arrays
unsigned deliver(chanend c_out, chanend ?c_spd_out, unsigned divide, chanend ?c_dig_rx, chanend ?c_adc)
{
unsigned sample;
#if NUM_USB_CHAN_OUT > 0
unsigned samplesOut[NUM_USB_CHAN_OUT];
#endif
#if NUM_USB_CHAN_IN > 0
unsigned samplesIn[NUM_USB_CHAN_IN];
unsigned samplesInPrev[NUM_USB_CHAN_IN];
#endif
unsigned tmp;
unsigned index;
#ifdef RAMP_CHECK
unsigned prev=0;
int started = 0;
#endif
unsigned test = 0;
#if NUM_USB_CHAN_IN > 0
for (int i=0;i<NUM_USB_CHAN_IN;i++)
{
samplesIn[i] = 0;
samplesInPrev[i] = 0;
}
#endif
outuint(c_out, 0);
/* Check for sample freq change or new samples from mixer*/
if(testct(c_out))
{
inct(c_out);
return inuint(c_out);
}
else
{
#ifndef MIXER // Interfaces straight to decouple()
(void) inuint(c_out);
#if NUM_USB_CHAN_IN > 0
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, samplesIn[i]);
}
#endif
#if NUM_USB_CHAN_OUT > 0
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
samplesOut[i] = inuint(c_out);
}
#endif
#else
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
int tmp = inuint(c_out);
#if defined(OUT_VOLUME_IN_MIXER) && defined(OUT_VOLUME_AFTER_MIX)
tmp<<=3;
#endif
samplesOut[i] = tmp;
}
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, samplesIn[i]);
}
#endif
}
#ifdef CODEC_SLAVE
/* Clear I2S port buffers */
clearbuf(p_lrclk);
#if (I2S_CHANS_DAC != 0)
for(int i = 0; i < I2S_WIRES_DAC; i++)
{
clearbuf(p_i2s_dac[i]);
}
#endif
#if (I2S_CHANS_ADC != 0)
for(int i = 0; i < I2S_WIRES_ADC; i++)
{
clearbuf(p_i2s_adc[i]);
}
#endif
if(divide == 1)
{
p_lrclk <: 0 @ tmp;
tmp += 30;
/* Prefill the ports so data starts to be input */
#if (I2S_CHANS_DAC != 0)
#pragma loop unroll
for(int i = 0; i < I2S_WIRES_DAC; i++)
{
p_i2s_dac[i] @ tmp <: 0;
}
#endif
p_lrclk @ tmp <: 0x7FFFFFFF;
#if (I2S_CHANS_ADC != 0)
p_i2s_adc[0] @ (tmp - 1) :> void;
#endif
#if (I2S_CHANS_ADC != 0)
#pragma loop unroll
for(int i = 0; i < I2S_WIRES_ADC; i++)
{
clearbuf(p_i2s_adc[i]);
}
#endif
}
else
{
clearbuf(p_bclk);
#if (I2S_CHANS_DAC != 0)
/* Prefill the ports so data is input in advance */
for(int i = 0; i < I2S_WIRES_DAC; i++)
{
p_i2s_dac[i] <: 0;
}
#endif
p_lrclk <: 0x7FFFFFFF;
p_bclk <: 0xAAAAAAAA;
p_bclk <: 0xAAAAAAAA;
}
#else
/* CODEC is master */
/* Wait for LRCLK edge */
p_lrclk when pinseq(0) :> void;
p_lrclk when pinseq(1) :> void;
p_lrclk when pinseq(0) :> void;
p_lrclk when pinseq(1) :> void;
p_lrclk when pinseq(0) :> void @ tmp;
tmp += 33;
#if (I2S_CHANS_DAC != 0)
#pragma loop unroll
for(int i = 0; i < I2S_WIRES_DAC; i++)
{
p_i2s_dac[i] @ tmp <: 0;
}
#endif
p_i2s_adc[0] @ tmp - 1 :> void;
#pragma loop unroll
for(int i = 0; i < I2S_WIRES_ADC; i++)
{
clearbuf(p_i2s_adc[i]);
}
/* TODO In master mode, the i/o loop assumes L/RCLK = 32bit clocks. We should check this every interation
* and resync if we got a bclk glitch */
#endif
/* Main Audio I/O loop */
while (1)
{
outuint(c_out, 0);
/* Check for sample freq change or new samples from mixer*/
if(testct(c_out))
{
inct(c_out);
return inuint(c_out);
}
else
{
#ifndef MIXER // Interfaces straight to decouple()
(void) inuint(c_out);
#if NUM_USB_CHAN_IN > 0
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, samplesIn[i]);
}
#endif
#if NUM_USB_CHAN_OUT > 0
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
samplesOut[i] = inuint(c_out);
}
#endif
#else
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
int tmp = inuint(c_out);
#if defined(OUT_VOLUME_IN_MIXER) && defined(OUT_VOLUME_AFTER_MIX)
tmp<<=3;
#endif
samplesOut[i] = tmp;
}
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, samplesIn[i]);
}
#endif
}
#if defined(SPDIF_RX) || defined(ADAT_RX)
inuint(c_dig_rx);
#endif
#ifdef SPDIF_RX
asm("ldw %0, dp[g_digData]":"=r"(samplesIn[SPDIF_RX_INDEX + 0]));
asm("ldw %0, dp[g_digData+4]":"=r"(samplesIn[SPDIF_RX_INDEX + 1]));
#endif
#ifdef ADAT_RX
asm("ldw %0, dp[g_digData+8]":"=r"(samplesIn[ADAT_RX_INDEX + 0]));
asm("ldw %0, dp[g_digData+12]":"=r"(samplesIn[ADAT_RX_INDEX+ 1]));
asm("ldw %0, dp[g_digData+16]":"=r"(samplesIn[ADAT_RX_INDEX + 2]));
asm("ldw %0, dp[g_digData+20]":"=r"(samplesIn[ADAT_RX_INDEX + 3]));
asm("ldw %0, dp[g_digData+24]":"=r"(samplesIn[ADAT_RX_INDEX + 4]));
asm("ldw %0, dp[g_digData+28]":"=r"(samplesIn[ADAT_RX_INDEX + 5]));
asm("ldw %0, dp[g_digData+32]":"=r"(samplesIn[ADAT_RX_INDEX + 6]));
asm("ldw %0, dp[g_digData+36]":"=r"(samplesIn[ADAT_RX_INDEX + 7]));
#endif
#if defined(SPDIF_RX) || defined(ADAT_RX)
/* Request digital data (with prefill) */
outuint(c_dig_rx, 0);
#endif
tmp = 0;
#pragma xta endpoint "i2s_output_l"
#if (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
#pragma loop unroll
for(int i = 0; i < I2S_CHANS_DAC; i+=2)
{
p_i2s_dac[tmp++] <: bitrev(samplesOut[i]); /* Output LEFT sample to DAC */
}
#endif
#ifdef CODEC_SLAVE
/* Generate clocks LR Clock low - LEFT */
switch (divide)
{
case 8:
/* LR clock delayed by one clock, This is so MSB is output on the falling edge of BCLK
* after the falling edge on which LRCLK was toggled. (see I2S spec) */
p_lrclk <: 0x80000000;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
break;
case 4:
p_lrclk <: 0x80000000;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
break;
case 2:
p_lrclk <: 0x80000000;
p_bclk <: 0xAAAAAAAA;
p_bclk <: 0xAAAAAAAA;
break;
case 1:
p_lrclk <: 0x80000000;
break;
}
#endif
#if (I2S_CHANS_ADC != 0)
/* Input prevous R sample into R in buffer */
index = 0;
#pragma loop unroll
for(int i = 1; i < I2S_CHANS_ADC; i += 2)
{
p_i2s_adc[index++] :> sample;
#if NUM_USB_CHAN_IN > 0
samplesIn[i] = bitrev(sample);
/* Store the previous left in left */
samplesIn[i-1] = samplesInPrev[i];
#endif
}
#endif
#if defined(SPDIF) && (NUM_USB_CHAN_OUT > 0)
outuint(c_spd_out, samplesOut[SPDIF_TX_INDEX]); /* Forward sample to SPDIF txt thread */
sample = samplesOut[SPDIF_TX_INDEX + 1];
outuint(c_spd_out, sample); /* Forward sample to SPDIF txt thread */
#ifdef RAMP_CHECK
sample >>= 8;
if (started<10000) {
if (sample == prev+1)
started++;
}
else
if (sample != prev+1 && sample != 0) {
printintln(prev);
printintln(sample);
printintln(prev-sample+1);
}
prev = sample;
#endif
#endif
tmp = 0;
#pragma xta endpoint "i2s_output_r"
#if (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
#pragma loop unroll
for(int i = 1; i < I2S_CHANS_DAC; i+=2)
{
p_i2s_dac[tmp++] <: bitrev(samplesOut[i]); /* Output RIGHT sample to DAC */
}
#endif
#ifdef CODEC_SLAVE
/* Clock out data (and LR clock) */
switch (divide)
{
case 8:
p_lrclk <: 0x7FFFFFFF;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
p_bclk <: 0xF0F0F0F0;
break;
case 4:
p_lrclk <: 0x7FFFFFFF;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
p_bclk <: 0xCCCCCCCC;
break;
case 2:
p_lrclk <: 0x7FFFFFFF;
p_bclk <: 0xAAAAAAAA;
p_bclk <: 0xAAAAAAAA;
break;
case 1:
p_lrclk <: 0x7FFFFFFF;
break;
}
#endif
#if (I2S_CHANS_ADC != 0)
/* Input previous L ADC sample */
index = 0;
#pragma loop unroll
for(int i = 1; i < I2S_CHANS_ADC; i += 2)
{
p_i2s_adc[index++] :> sample;
#if NUM_USB_CHAN_IN > 0
samplesInPrev[i] = bitrev(sample);
#endif
}
#ifdef SU1_ADC
{
unsigned x;
x = inuint(c_adc);
inct(c_adc);
asm("stw %0, dp[g_adcVal]"::"r"(x));
}
#endif
#endif
}
return 0;
}
/* This function is a dummy version of the deliver thread that does not
connect to the codec ports. It is used during DFU reset. */
static unsigned dummy_deliver(chanend c_out) {
while (1)
{
outuint(c_out, 0);
/* Check for sample freq change or new samples from mixer*/
if(testct(c_out))
{
inct(c_out);
return inuint(c_out);
}
else
{
#ifndef MIXER // Interfaces straight to decouple()
(void) inuint(c_out);
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, 0);
}
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
(void) inuint(c_out);
}
#else
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_OUT; i++)
{
(void) inuint(c_out);
}
#pragma loop unroll
for(int i = 0; i < NUM_USB_CHAN_IN; i++)
{
outuint(c_out, 0);
}
#endif
}
}
return 0;
}
#define SAMPLE_RATE 200000
#define NUMBER_CHANNELS 1
#define NUMBER_SAMPLES 100
#define NUMBER_WORDS ((NUMBER_SAMPLES * NUMBER_CHANNELS+1)/2)
#define SAMPLES_PER_PRINT 1
void audio(chanend c_mix_out, chanend ?c_dig_rx, chanend ?c_config, chanend ?c)
{
#ifdef SPDIF
chan c_spdif_out;
#endif
unsigned curSamFreq = DEFAULT_FREQ;
unsigned mClk;
unsigned divide;
unsigned firstRun = 1;
#ifdef SU1_ADC
/* Setup galaxian ADC */
unsigned data[1], channel;
int r;
unsigned int vals[NUMBER_WORDS];
int cnt = 0;
int div;
unsigned val = 0;
int val2 = 0;
int adcOk = 0;
/* Enable adc on channel */
enable_xs1_su_adc_input(0, c);
/* General ADC control (enabled, 1 samples per packet, 32 bits per sample) */
data[0] = 0x10201;
data[0] = 0x30101;
r = write_periph_32(xs1_su, 2, 0x20, 1, data);
/* ADC needs a few clocks before it starts pumping out samples */
for(int i = 0; i< 10; i++)
{
p_lrclk <: val;
val = ~val;
{
timer t;
unsigned time;
t :> time;
t when timerafter(time+1000):> void;
}
}
#endif
#ifdef SPDIF
SpdifTransmitPortConfig(p_spdif_tx, clk_mst_spd, p_mclk);
#endif
/* Initialise master clock generation */
ClockingInit(c_config);
/* Perform required CODEC/ADC/DAC initialisation */
CodecInit(c_config);
{
}
while(1)
{
/* Calculate what master clock we should be using */
if ((curSamFreq % 22050) == 0)
{
mClk = MCLK_441;
}
else if ((curSamFreq % 24000) == 0)
{
mClk = MCLK_48;
}
/* Calculate divide required for bit clock e.g. 11.289600 / (176400 * 64) = 1 */
divide = mClk / ( curSamFreq * 64 );
/* Configure clocking for required master clock */
ClockingConfig(mClk, c_config);
if(!firstRun)
{
/* TODO wait for good mclk instead of delay */
/* No delay for DFU modes */
if ((curSamFreq != AUDIO_REBOOT_FROM_DFU) && (curSamFreq != AUDIO_STOP_FOR_DFU))
{
timer t;
unsigned time;
t :> time;
t when timerafter(time+AUDIO_PLL_LOCK_DELAY) :> void;
/* Handshake back */
outct(c_mix_out, XS1_CT_END);
}
}
firstRun = 0;
/* Configure CODEC/DAC/ADC for SampleFreq/MClk */
CodecConfig(curSamFreq, mClk, c_config);
/* Configure audio ports */
ConfigAudioPorts(divide);
par
{
#ifdef SPDIF
{
set_thread_fast_mode_on();
SpdifTransmit(p_spdif_tx, c_spdif_out);
}
#endif
{
#ifdef SPDIF
/* Communicate master clock and sample freq to S/PDIF thread */
outuint(c_spdif_out, curSamFreq);
outuint(c_spdif_out, mClk);
#endif
curSamFreq = deliver(c_mix_out,
#ifdef SPDIF
c_spdif_out,
#else
null,
#endif
divide, c_dig_rx, c);
// Currently no more audio will happen after this point
if (curSamFreq == AUDIO_STOP_FOR_DFU)
{
outct(c_mix_out, XS1_CT_END);
while (1)
{
curSamFreq = dummy_deliver(c_mix_out);
if (curSamFreq == AUDIO_START_FROM_DFU)
{
outct(c_mix_out, XS1_CT_END);
break;
}
}
}
#ifdef SPDIF
/* Notify S/PDIF thread of impending new freq... */
outct(c_spdif_out, XS1_CT_END);
#endif
}
}
}
}