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

Memory optimisation of I2S DS3 integration

Browse Source
This commit is contained in:
Sam Chesney
2016-12-06 17:29:11 +00:00
parent c5eb740637
commit bcc9c23a71
1 changed files with 54 additions and 78 deletions
Download Patch File Download Diff File Expand all files Collapse all files

132
module_usb_audio/audio_io/audio_io.xc
View File

@@ -41,10 +41,6 @@
#error "Unsupported I2S downsampling configuration"
#endif
#if (I2S_DOWNSAMPLE_FACTOR > 1) && (I2S_CHANS_DAC > 2)
#error "I2S downsampling only supports stereo I2S"
#endif
#if (NUM_USB_CHAN_IN && (NUM_USB_CHAN_IN < (I2S_CHANS_ADC + NUM_PDM_MICS)))
#error "Not enough USB input channels to support number of I2S and PDM inputs"
#endif
@@ -57,17 +53,18 @@
static unsigned samplesOut[32];
/* Two buffers for ADC data to allow for DAC and ADC ports being offset */
static unsigned samplesIn[2][I2S_DOWNSAMPLE_FACTOR][32];
static unsigned samplesIn[2][32];
static int downsamplingCounter = 0;
#if (I2S_DOWNSAMPLE_FACTOR > 1)
#include "src.h"
/* [Number of I2S channels][Number of samples/phases][Taps per phase] */
static union ds3Data
{
long long doubleWordAlignmentEnsured;
int32_t delayLine[PDM_MIC_INDEX][I2S_DOWNSAMPLE_FACTOR][24];
/* [Number of I2S channels][Number of samples/phases][Taps per phase] */
int32_t delayLine[I2S_CHANS_DAC][I2S_DOWNSAMPLE_FACTOR][24];
} ds3Data;
static int64_t ds3Sum[I2S_CHANS_DAC];
#endif
#if (DSD_CHANS_DAC != 0)
@@ -299,18 +296,18 @@ static inline unsigned DoSampleTransfer(chanend c_out, const int readBuffNo, con
#else
inuint(c_out);
#endif
UserBufferManagement(samplesOut, samplesIn[readBuffNo][downsamplingCounter], i_audMan);
UserBufferManagement(samplesOut, samplesIn[readBuffNo], i_audMan);
#if NUM_USB_CHAN_IN > 0
#pragma loop unroll
for(int i = 0; i < I2S_CHANS_ADC; i++)
{
outuint(c_out, samplesIn[readBuffNo][downsamplingCounter][i]);
outuint(c_out, samplesIn[readBuffNo][i]);
}
#pragma loop unroll
for (int i = PDM_MIC_INDEX; i < (NUM_PDM_MICS + PDM_MIC_INDEX); i++)
{
outuint(c_out, samplesIn[readBuffNo][downsamplingCounter][i]);
outuint(c_out, samplesIn[readBuffNo][i]);
}
#endif
}
@@ -504,8 +501,7 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
#endif
#if (I2S_DOWNSAMPLE_FACTOR > 1)
const int ds3DelayLine_zero = {0};
memcpy(&ds3Data.delayLine, &ds3DelayLine_zero, sizeof ds3DelayLine_zero);
memset(&ds3Data.delayLine, 0, sizeof ds3Data);
#endif
unsigned command = DoSampleTransfer(c_out, readBuffNo, underflowWord, i_audMan);
@@ -639,6 +635,12 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
else
#endif
{
#if (I2S_DOWNSAMPLE_FACTOR > 1)
if (0 == downsamplingCounter)
{
memset(&ds3Sum, 0, sizeof ds3Sum);
}
#endif
#if (I2S_CHANS_ADC != 0)
/* Input previous L sample into L in buffer */
index = 0;
@@ -655,41 +657,28 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
asm volatile("in %0, res[%1]" : "=r"(sample) : "r"(p_i2s_adc[index++]));
/* Note the use of readBuffNo changes based on frameCount */
samplesIn[buffIndex][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 0, 2, 4.. on each line.
}
samplesIn[buffIndex][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 0, 2, 4.. on each line.
#if (I2S_DOWNSAMPLE_FACTOR > 1)
switch (downsamplingCounter)
{
case 0:
samplesIn[readBuffNo][2][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] = 0;
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_sample(
samplesIn[readBuffNo][2][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[0][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
break;
case 1:
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_sample(
samplesIn[readBuffNo][0][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[0][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
break;
case 2:
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] =
if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
{
samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_final_sample(
samplesIn[readBuffNo][1][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[0][downsamplingCounter],
ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i],
ds3Data.delayLine[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][downsamplingCounter][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
break;
default:
unreachable("downsamplingCounter out of range");
break;
}
samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
}
else
{
ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] =
src_ds3_voice_add_sample(
ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i],
ds3Data.delayLine[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))]);
}
#endif // (I2S_DOWNSAMPLE_FACTOR > 1)
}
#endif
@@ -771,7 +760,7 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
#pragma loop unroll
for(int i = PDM_MIC_INDEX; i < (NUM_PDM_MICS + PDM_MIC_INDEX); i++)
{
c_pdm_pcm :> samplesIn[readBuffNo][downsamplingCounter][i];
c_pdm_pcm :> samplesIn[readBuffNo][i];
}
}
}
@@ -788,42 +777,29 @@ unsigned static deliver(chanend c_out, chanend ?c_spd_out,
unsigned sample;
asm volatile("in %0, res[%1]" : "=r"(sample) : "r"(p_i2s_adc[index++]));
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 1, 3, 5.. on each line.
samplesIn[buffIndex][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i] = bitrev(sample); // channels 1, 3, 5.. on each line.
#if (I2S_DOWNSAMPLE_FACTOR > 1)
if ((I2S_DOWNSAMPLE_FACTOR - 1) == downsamplingCounter)
{
samplesIn[buffIndex][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i] =
src_ds3_voice_add_final_sample(
ds3Sum[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i],
ds3Data.delayLine[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i]);
}
else
{
ds3Sum[((frameCount-2)&(I2S_CHANS_PER_FRAME-1))+i] =
src_ds3_voice_add_sample(
ds3Sum[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i],
ds3Data.delayLine[((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[readBuffNo][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))+i]);
}
#endif // (I2S_DOWNSAMPLE_FACTOR > 1)
}
#if (I2S_DOWNSAMPLE_FACTOR > 1)
switch (downsamplingCounter)
{
case 0:
samplesIn[readBuffNo][2][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))] = 0;
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_sample(
samplesIn[readBuffNo][2][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[1][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))]);
break;
case 1:
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_sample(
samplesIn[readBuffNo][0][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[1][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))]);
break;
case 2:
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))] =
src_ds3_voice_add_final_sample(
samplesIn[readBuffNo][1][((frameCount-2)&(I2S_CHANS_PER_FRAME-1))],
ds3Data.delayLine[1][downsamplingCounter],
src_ds3_voice_coefs[downsamplingCounter],
samplesIn[buffIndex][downsamplingCounter][((frameCount-1)&(I2S_CHANS_PER_FRAME-1))]);
break;
default:
unreachable("downsamplingCounter out of range");
break;
}
#endif // (I2S_DOWNSAMPLE_FACTOR > 1)
#endif
#ifndef CODEC_MASTER