dengweijie/lib_xua
1
0
Fork 0
forked from PAWPAW-Mirror/lib_xua
Code Pull Requests Activity

First cut at reunifying master/slave i2s loops

Browse Source
This commit is contained in:
xross
2018-03-08 20:07:30 +00:00
parent 6824657b4f
commit 961d890b2c
4 changed files with 86 additions and 407 deletions
Download Patch File Download Diff File Expand all files Collapse all files

11
lib_xua/api/xua_conf_default.h
View File

@@ -1453,3 +1453,14 @@ enum USBEndpointNumber_Out
#define MIN_FREQ_44 (((44100*512)/((48000 * 512)/MIN_FREQ))*2)
#endif
/* Internal define for port declaration */
#if CODEC_MASTER
#define _XUA_CLK_DIR in
#else
#define _XUA_CLK_DIR out
#endif
#if (CODEC_MASTER == 1) && (DSD_CHANS_DAC != 0)
#error CODEC_MASTER with DSD is currently unsupported
#endif

6
lib_xua/api/xua_conf_full.h
View File

@@ -8,10 +8,6 @@
#include "xua_conf_default.h"
#if CODEC_MASTER
#define _XUA_CLK_DIR in
#else
#define _XUA_CLK_DIR out
#endif
#endif

4
lib_xua/src/core/audiohub/init_ports.h
View File

@@ -59,9 +59,9 @@ static inline void InitPorts_master(unsigned divide, buffered out port:32 p_lrcl
}
#endif
}
#endif
#if CODEC_MASTER
#else
static inline void InitPorts_slave(unsigned divide, buffered in port:32 p_lrclk, buffered in port:32 p_bclk)
{
#if (I2S_CHANS_ADC != 0 || I2S_CHANS_DAC != 0)

472
lib_xua/src/core/audiohub/xua_audiohub.xc
View File

@@ -212,10 +212,60 @@ static inline unsigned DoSampleTransfer(chanend c_out, const int readBuffNo, con
return 0;
}
static inline int HandleSampleClock(int frameCount, buffered _XUA_CLK_DIR port:32 p_lrclk)
{
#if CODEC_MASTER
unsigned syncError = 0;
unsigned lrval = 0;
p_lrclk :> lrval;
if(I2S_MODE_TDM)
{
/* Only check for the rising edge of frame sync being in the right place because falling edge timing not specified */
if (frameCount == (I2S_CHANS_PER_FRAME-1))
{
lrval &= 0xc0000000; // Mask off last two (MSB) frame clock bits which are the most recently sampled
syncError += (lrval != 0x80000000); // We need MSB = 1 and MSB-1 = 0 to signify rising edge
}
else
{
/* We do not check this part of the frame because TDM frame sync falling egde timing
* is not defined. We only care about rising edge which is checked in first half of frame */
}
}
else
{
if(frameCount == 0)
syncError += (lrval != 0x80000000);
else
syncError += (lrval != 0x7FFFFFFF);
}
return syncError;
#else
if(I2S_MODE_TDM)
{
if(frameCount == (I2S_CHANS_PER_FRAME-1))
p_lrclk <: 0x80000000;
else
p_lrclk <: 0x00000000;
}
else
{
if(frameCount == 0)
p_lrclk <: 0x80000000;
else
p_lrclk <: 0x7fffffff;
}
return 0;
#endif
}
#if (CODEC_MASTER == 0)
#pragma unsafe arrays
unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
unsigned static AudioHub_MainLoop(chanend ?c_out, chanend ?c_spd_out
#ifdef ADAT_TX
, chanend c_adat_out
, unsigned adatSmuxMode
@@ -227,7 +277,7 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#if (NUM_PDM_MICS > 0)
, chanend c_pdm_pcm
#endif
, buffered out port:32 p_lrclk, buffered out port:32 p_bclk
, buffered _XUA_CLK_DIR port:32 p_lrclk, buffered _XUA_CLK_DIR port:32 p_bclk
)
{
/* Since DAC and ADC buffered ports off by one sample we buffer previous ADC frame */
@@ -319,11 +369,19 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
return command;
}
InitPorts_master(divide, p_lrclk, p_bclk);
/* Main Audio I/O loop */
while (1)
{
unsigned syncError = 0;
#if CODEC_MASTER
InitPorts_slave(divide, p_lrclk, p_bclk);
#else
InitPorts_master(divide, p_lrclk, p_bclk);
#endif
/* Note we always expect syncError to be 0 when we are master */
while(!syncError)
{
#if (DSD_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT > 0)
if(dsdMode == DSD_MODE_NATIVE)
@@ -381,15 +439,8 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
}
#endif
/* 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) */
/* Generate clocks LR Clock low - LEFT */
#if (I2S_CHANS_ADC != 0 || I2S_CHANS_DAC != 0)
#if I2S_MODE_TDM
p_lrclk <: 0x00000000;
#else
p_lrclk <: 0x80000000;
#endif
syncError += HandleSampleClock(frameCount, p_lrclk);
#endif
#pragma xta endpoint "i2s_output_l"
@@ -477,6 +528,8 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#endif
}
frameCount++;
#if (I2S_CHANS_ADC != 0)
index = 0;
/* Channels 0, 2, 4.. on each line */
@@ -487,7 +540,7 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
unsigned sample;
asm volatile("in %0, res[%1]" : "=r"(sample) : "r"(p_i2s_adc[index++]));
sample = bitrev(sample);
int chanIndex = ((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i; // channels 1, 3, 5.. on each line.
int chanIndex = ((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i; // channels 1, 3, 5.. on each line.
#if (AUD_TO_USB_RATIO > 1 && !I2S_DOWNSAMPLE_MONO_IN)
if ((AUD_TO_USB_RATIO - 1) == audioToUsbRatioCounter)
{
@@ -514,14 +567,7 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#endif
#if (I2S_CHANS_ADC != 0 || I2S_CHANS_DAC != 0)
#if I2S_MODE_TDM
if(frameCount == (I2S_CHANS_PER_FRAME-2))
p_lrclk <: 0x80000000;
else
p_lrclk <: 0x00000000;
#else
p_lrclk <: 0x7FFFFFFF;
#endif
syncError += HandleSampleClock(frameCount, p_lrclk);
#endif
index = 0;
@@ -529,7 +575,7 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#if (I2S_CHANS_DAC != 0)
/* Output "odd" channel to DAC (i.e. right) */
#pragma loop unroll
for(int i = 1; i < I2S_CHANS_DAC; i+=I2S_CHANS_PER_FRAME)
for(int i = 0; i < I2S_CHANS_DAC; i+=I2S_CHANS_PER_FRAME)
{
#if (AUD_TO_USB_RATIO > 1)
if (audioToUsbRatioCounter == 0)
@@ -558,7 +604,7 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#if I2S_MODE_TDM
/* Increase frameCount by 2 since we have output two channels (per data line) */
frameCount+=2;
frameCount+=1;
if(frameCount == I2S_CHANS_PER_FRAME)
#endif
{
@@ -592,379 +638,8 @@ unsigned static deliver_master(chanend ?c_out, chanend ?c_spd_out
#pragma xta endpoint "deliver_return"
return 0;
}
#endif
#if (CODEC_MASTER == 1)
/* I2S delivery thread */
#pragma unsafe arrays
unsigned static deliver_slave(chanend ?c_out, chanend ?c_spd_out
#ifdef ADAT_TX
, chanend c_adat_out
, unsigned adatSmuxMode
#endif
, unsigned divide, unsigned curSamFreq
#if(SPDIF_RX == 1) || (ADAT_RX == 1)
, chanend c_dig_rx
#endif
#if (NUM_PDM_MICS > 0)
, chanend c_pdm_pcm
#endif
, buffered in port:32 p_lrclk, buffered in port:32 p_bclk
)
{
/* Since DAC and ADC buffered ports off by one sample we buffer previous ADC frame */
unsigned readBuffNo = 0;
unsigned index;
#ifdef RAMP_CHECK
unsigned prev=0;
int started = 0;
#endif
int firstIteration = 1;
unsigned underflowWord = 0;
unsigned frameCount = 0;
#ifdef ADAT_TX
adatCounter = 0;
#endif
unsigned audioToUsbRatioCounter = 0;
#if (NUM_PDM_MICS > 0)
unsigned audioToMicsRatioCounter = 0;
#endif
#if (AUD_TO_USB_RATIO > 1)
union i2sInDs3
{
long long doubleWordAlignmentEnsured;
int32_t delayLine[I2S_DOWNSAMPLE_CHANS_IN][SRC_FF3V_FIR_NUM_PHASES][SRC_FF3V_FIR_TAPS_PER_PHASE];
} i2sInDs3;
memset(&i2sInDs3.delayLine, 0, sizeof i2sInDs3.delayLine);
int64_t i2sInDs3Sum[I2S_DOWNSAMPLE_CHANS_IN];
union i2sOutUs3
{
long long doubleWordAlignmentEnsured;
int32_t delayLine[I2S_CHANS_DAC][SRC_FF3V_FIR_TAPS_PER_PHASE];
} i2sOutUs3;
memset(&i2sOutUs3.delayLine, 0, sizeof i2sOutUs3.delayLine);
#endif /* (AUD_TO_USB_RATIO > 1) */
#if ((DEBUG_MIC_ARRAY == 1) && (NUM_PDM_MICS > 0))
/* Get initial samples from PDM->PCM converter to avoid stalling the decimators */
c_pdm_pcm <: 1;
master
{
#pragma loop unroll
for(int i = PDM_MIC_INDEX; i < (NUM_PDM_MICS + PDM_MIC_INDEX); i++)
{
c_pdm_pcm :> samplesIn[readBuffNo][i];
}
}
#endif // ((DEBUG_MIC_ARRAY == 1) && (NUM_PDM_MICS > 0))
UserBufferManagementInit();
unsigned command = DoSampleTransfer(c_out, readBuffNo, underflowWord);
// Reinitialise user state before entering the main loop
UserBufferManagementInit();
#ifdef ADAT_TX
unsafe{
//TransferAdatTxSamples(c_adat_out, samplesOut, adatSmuxMode, 0);
volatile unsigned * unsafe samplePtr = &samplesOut[ADAT_TX_INDEX];
outuint(c_adat_out, (unsigned) samplePtr);
}
#endif
if(command)
{
return command;
}
/* Main Audio I/O loop */
while (1)
{
/* In CODEC master mode, the I/O loop assumes L/RCLK = 32bit clocks.
* Check this every iteration and resync if we get a bclk glitch.
*/
unsigned lrval;
unsigned syncError = 0;
InitPorts_slave(divide, p_lrclk, p_bclk);
while (!syncError)
{
#if (I2S_CHANS_ADC != 0)
#if (AUD_TO_USB_RATIO > 1)
if (0 == audioToUsbRatioCounter)
{
memset(&i2sInDs3Sum, 0, sizeof i2sInDs3Sum);
}
#endif /* (AUD_TO_USB_RATIO > 1) */
/* Input previous L sample into L in buffer */
index = 0;
/* First input (i.e. frameCount == 0) we read last ADC channel of previous frame.. */
unsigned buffIndex = (frameCount > 1) ? !readBuffNo : readBuffNo;
#pragma loop unroll
/* First time around we get channel 7 of TDM8 */
for(int i = 0; i < I2S_CHANS_ADC; i+=I2S_CHANS_PER_FRAME)
{
// p_i2s_adc[index++] :> sample;
// Manual IN instruction since compiler generates an extra setc per IN (bug #15256)
unsigned sample;
asm volatile("in %0, res[%1]" : "=r"(sample) : "r"(p_i2s_adc[index++]));
sample = bitrev(sample);
int chanIndex = ((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i; // channels 0, 2, 4.. on each line.
#if (AUD_TO_USB_RATIO > 1)
if ((AUD_TO_USB_RATIO - 1) == audioToUsbRatioCounter)
{
samplesIn[buffIndex][chanIndex] =
src_ds3_voice_add_final_sample(
i2sInDs3Sum[chanIndex],
i2sInDs3.delayLine[chanIndex][audioToUsbRatioCounter],
src_ff3v_fir_coefs[audioToUsbRatioCounter],
sample);
}
else
{
i2sInDs3Sum[chanIndex] =
src_ds3_voice_add_sample(
i2sInDs3Sum[chanIndex],
i2sInDs3.delayLine[chanIndex][audioToUsbRatioCounter],
src_ff3v_fir_coefs[audioToUsbRatioCounter],
sample);
}
#else
samplesIn[buffIndex][chanIndex] = sample;
#endif /* (AUD_TO_USB_RATIO > 1) */
}
#endif //(I2S_CHANS_ADC != 0)
#if (I2S_CHANS_ADC != 0 || I2S_CHANS_DAC != 0)
/* LR Clock sync check */
p_lrclk :> lrval;
#endif
if(I2S_MODE_TDM)
{
/* Only check for the rising edge of frame sync being in the right place because falling edge timing not specified */
if (frameCount == (I2S_CHANS_PER_FRAME-1))
{
lrval &= 0xc0000000; // Mask off last two (MSB) frame clock bits which are the most recently sampled
syncError += (lrval != 0x80000000); // We need MSB = 1 and MSB-1 = 0 to signify rising edge
}
}
else
{
syncError += (lrval != 0x80000000);
}
#pragma xta endpoint "i2s_output_l"
#if (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
index = 0;
#pragma loop unroll
/* Output "even" channel to DAC (i.e. left) */
for(int i = 0; i < I2S_CHANS_DAC; i+=I2S_CHANS_PER_FRAME)
{
#if (AUD_TO_USB_RATIO > 1)
if (0 == audioToUsbRatioCounter)
{
samplesOut[frameCount+i] = src_us3_voice_input_sample(i2sOutUs3.delayLine[i],
src_ff3v_fir_coefs[2],
samplesOut[frameCount+i]);
}
else /* audioToUsbRatioCounter == 1 or 2 */
{
samplesOut[frameCount+i] = src_us3_voice_get_next_sample(i2sOutUs3.delayLine[i],
src_ff3v_fir_coefs[2-audioToUsbRatioCounter]);
}
#endif /* (AUD_TO_USB_RATIO > 1) */
p_i2s_dac[index++] <: bitrev(samplesOut[frameCount +i]);
}
#endif // (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
#ifdef ADAT_TX
TransferAdatTxSamples(c_adat_out, samplesOut, adatSmuxMode, 1);
#endif
if(frameCount == 0)
{
#if (SPDIF_RX == 1) || (ADAT_RX == 1)
/* Sync with clockgen */
inuint(c_dig_rx);
/* Note, digi-data we just store in samplesIn[readBuffNo] - we only double buffer the I2S input data */
#endif
#if (SPDIF_RX == 1)
asm("ldw %0, dp[g_digData]" :"=r"(samplesIn[readBuffNo][SPDIF_RX_INDEX + 0]));
asm("ldw %0, dp[g_digData+4]":"=r"(samplesIn[readBuffNo][SPDIF_RX_INDEX + 1]));
#endif
#if (ADAT_RX == 1)
asm("ldw %0, dp[g_digData+8]" :"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX]));
asm("ldw %0, dp[g_digData+12]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 1]));
asm("ldw %0, dp[g_digData+16]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 2]));
asm("ldw %0, dp[g_digData+20]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 3]));
asm("ldw %0, dp[g_digData+24]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 4]));
asm("ldw %0, dp[g_digData+28]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 5]));
asm("ldw %0, dp[g_digData+32]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 6]));
asm("ldw %0, dp[g_digData+36]":"=r"(samplesIn[readBuffNo][ADAT_RX_INDEX + 7]));
#endif
#if (SPDIF_RX== 1) || (ADAT_RX == 1)
/* Request digital data (with prefill) */
outuint(c_dig_rx, 0);
#endif
#if ((XUA_SPDIF_TX_EN) && (NUM_USB_CHAN_OUT > 0))
outuint(c_spd_out, samplesOut[SPDIF_TX_INDEX]); /* Forward sample to S/PDIF Tx thread */
unsigned sample = samplesOut[SPDIF_TX_INDEX + 1];
outuint(c_spd_out, sample); /* Forward sample to S/PDIF Tx thread */
#endif
#if (NUM_PDM_MICS > 0)
if ((AUD_TO_MICS_RATIO - 1) == audioToMicsRatioCounter)
{
/* Get samples from PDM->PCM converter */
c_pdm_pcm <: 1;
master
{
#pragma loop unroll
for(int i = PDM_MIC_INDEX; i < (NUM_PDM_MICS + PDM_MIC_INDEX); i++)
{
c_pdm_pcm :> samplesIn[readBuffNo][i];
}
}
audioToMicsRatioCounter = 0;
}
else
{
++audioToMicsRatioCounter;
}
#endif
}
#if (I2S_CHANS_ADC != 0)
index = 0;
/* Channels 0, 2, 4.. on each line */
#pragma loop unroll
for(int i = 0; i < I2S_CHANS_ADC; i += I2S_CHANS_PER_FRAME)
{
/* Manual IN instruction since compiler generates an extra setc per IN (bug #15256) */
unsigned sample;
asm volatile("in %0, res[%1]" : "=r"(sample) : "r"(p_i2s_adc[index++]));
sample = bitrev(sample);
int chanIndex = ((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i; // channels 1, 3, 5.. on each line.
#if (AUD_TO_USB_RATIO > 1 && !I2S_DOWNSAMPLE_MONO_IN)
if ((AUD_TO_USB_RATIO - 1) == audioToUsbRatioCounter)
{
samplesIn[buffIndex][chanIndex] =
src_ds3_voice_add_final_sample(
i2sInDs3Sum[chanIndex],
i2sInDs3.delayLine[chanIndex][audioToUsbRatioCounter],
src_ff3v_fir_coefs[audioToUsbRatioCounter],
sample);
}
else
{
i2sInDs3Sum[chanIndex] =
src_ds3_voice_add_sample(
i2sInDs3Sum[chanIndex],
i2sInDs3.delayLine[chanIndex][audioToUsbRatioCounter],
src_ff3v_fir_coefs[audioToUsbRatioCounter],
sample);
}
#else
samplesIn[buffIndex][chanIndex] = sample;
#endif /* (AUD_TO_USB_RATIO > 1) && !I2S_DOWNSAMPLE_MONO_IN */
}
#endif //(I2S_CHANS_ADC != 0)
#if (I2S_CHANS_ADC != 0 || I2S_CHANS_DAC != 0)
/* LR Clock sync check */
p_lrclk :> lrval;
#endif
if(I2S_MODE_TDM)
{
/* Do nothing */
// We do not check this part of the frame because TDM frame sync falling egde timing
// is not defined. We only care about rising edge which is checked in first half of frame
}
else
{
syncError += (lrval != 0x7fffffff);
}
index = 0;
#pragma xta endpoint "i2s_output_r"
#if (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
/* Output "odd" channel to DAC (i.e. right) */
#pragma loop unroll
for(int i = 1; i < I2S_CHANS_DAC; i+=I2S_CHANS_PER_FRAME)
{
#if (AUD_TO_USB_RATIO > 1)
if (audioToUsbRatioCounter == 0)
{
samplesOut[frameCount+i] = src_us3_voice_input_sample(i2sOutUs3.delayLine[i],
src_ff3v_fir_coefs[2],
samplesOut[frameCount+i]);
}
else
{ /* audioToUsbRatioCounter is 1 or 2 */
samplesOut[frameCount+i] = src_us3_voice_get_next_sample(i2sOutUs3.delayLine[i],
src_ff3v_fir_coefs[2-audioToUsbRatioCounter]);
}
#endif /* (AUD_TO_USB_RATIO > 1) */
p_i2s_dac[index++] <: bitrev(samplesOut[frameCount + i]);
}
#endif // (I2S_CHANS_DAC != 0) && (NUM_USB_CHAN_OUT != 0)
#if I2S_MODE_TDM
/* Increase frameCount by 2 since we have output two channels (per data line) */
frameCount+=2;
if(frameCount == I2S_CHANS_PER_FRAME)
#endif
{
if ((AUD_TO_USB_RATIO - 1) == audioToUsbRatioCounter)
{
/* Do samples transfer */
/* The below looks a bit odd but forces the compiler to inline twice */
unsigned command;
if(readBuffNo)
command = DoSampleTransfer(c_out, 1, underflowWord);
else
command = DoSampleTransfer(c_out, 0, underflowWord);
if(command)
{
return command;
}
/* Reset frame counter and flip the ADC buffer */
audioToUsbRatioCounter = 0;
frameCount = 0;
readBuffNo = !readBuffNo;
}
else
{
++audioToUsbRatioCounter;
}
}
}
}
#pragma xta endpoint "deliver_return"
return 0;
}
#endif //CODEC_MASTER
#if (XUA_SPDIF_TX_EN) && (SPDIF_TX_TILE != AUDIO_IO_TILE)
void SpdifTxWrapper(chanend c_spdif_tx)
{
@@ -1301,11 +976,8 @@ void XUA_AudioHub(chanend ?c_aud, clock ?clk_audio_mclk, clock ?clk_audio_bclk,
outuint(c_adat_out, adatMultiple);
outuint(c_adat_out, adatSmuxMode);
#endif
#if CODEC_MASTER
command = deliver_slave(c_aud
#else
command = deliver_master(c_aud
#endif
command = AudioHub_MainLoop(c_aud
#if (XUA_SPDIF_TX_EN)
, c_spdif_out
#else