hexianglong/snapclient
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

read audio data as 32 bit in dsp_processor

Browse Source
This commit is contained in:
raul
2022-12-04 17:12:01 +01:00
Unverified
parent ec1a7f5ee8
commit 9d0a17552c
1 changed files with 9 additions and 17 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
components/dsp_processor/dsp_processor.c
View File

@@ -44,6 +44,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
int16_t len = chunk_size / 4;
int16_t valint;
uint16_t i;
volatile uint32_t *audio_tmp = (uint32_t *)audio; //volatile needed to ensure 32 bit access
if ((sbuffer0 == NULL) || (sbufout0 == NULL) || (sbuftmp0 == NULL))
{
@@ -66,8 +67,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 1] << 8)
+ audio[i * 4 + 0]))
* ((float)((int16_t)(audio_tmp[i] & 0xFFFF)))
/ 32768;
}
BIQUAD (sbuffer0, sbufout0, len, bq[6].coeffs, bq[6].w);
@@ -75,17 +75,14 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 0] = (valint & 0x00ff);
audio[i * 4 + 1] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF0000) + (uint32_t)valint;
}
// channel 1
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 3] << 8)
+ audio[i * 4 + 2]))
* ((float)((int16_t)((audio_tmp[i] & 0xFFFF0000) >> 16)))
/ 32768;
}
BIQUAD (sbuffer0, sbufout0, len, bq[7].coeffs, bq[7].w);
@@ -93,8 +90,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 2] = (valint & 0x00ff);
audio[i * 4 + 3] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF) + ((uint32_t)valint << 16);
}
}
break;
@@ -105,8 +101,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 1] << 8)
+ audio[i * 4 + 0]))
* ((float)((int16_t)(audio_tmp[i] & 0xFFFF)))
/ 32768;
}
BIQUAD (sbuffer0, sbuftmp0, len, bq[0].coeffs, bq[0].w);
@@ -115,16 +110,14 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 0] = (valint & 0x00ff);
audio[i * 4 + 1] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF0000) + (uint32_t)valint;
}
// Process audio ch1 HIGH PASS FILTER
for (i = 0; i < len; i++)
{
sbuffer0[i] = dynamic_vol * 0.5
* ((float)((int16_t) (audio[i * 4 + 3] << 8)
+ audio[i * 4 + 2]))
* ((float)((int16_t)((audio_tmp[i] & 0xFFFF0000) >> 16)))
/ 32768;
}
BIQUAD (sbuffer0, sbuftmp0, len, bq[2].coeffs, bq[2].w);
@@ -133,8 +126,7 @@ dsp_processor (char *audio, size_t chunk_size, dspFlows_t dspFlow)
for (i = 0; i < len; i++)
{
valint = (int16_t) (sbufout0[i] * 32768);
audio[i * 4 + 2] = (valint & 0x00ff);
audio[i * 4 + 3] = ((valint & 0xff00) >> 8);
audio_tmp[i] = (audio_tmp[i]&0xFFFF) + ((uint32_t)valint << 16);
}
}
break;