fixed point resampling option

This commit is contained in:
philippe44
2019-06-26 21:31:50 -07:00
parent 375a5aec2a
commit a75f1f0cd5
11 changed files with 272 additions and 11 deletions

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@@ -9,6 +9,7 @@ COMPONENT_ADD_LDFLAGS=-l$(COMPONENT_NAME) \
$(COMPONENT_PATH)/lib/libvorbisidec.a \
$(COMPONENT_PATH)/lib/libogg.a \
$(COMPONENT_PATH)/lib/libalac.a \
$(COMPONENT_PATH)/lib/libresample16.a \
$(COMPONENT_PATH)/lib/libsoxr.a

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@@ -0,0 +1,85 @@
/*
* FILE: resample16.h
*
* The configuration constants below govern
* the number of bits in the input sample and filter coefficients, the
* number of bits to the right of the binary-point for fixed-point math, etc.
*
*/
/* Conversion constants */
#define Nhc 8
#define Na 7
#define Np (Nhc+Na)
#define Npc (1<<Nhc)
#define Amask ((1<<Na)-1)
#define Pmask ((1<<Np)-1)
#define Nh 16
#define Nb 16
#define Nhxn 14
#define Nhg (Nh-Nhxn)
#define NLpScl 13
/* Description of constants:
*
* Npc - is the number of look-up values available for the lowpass filter
* between the beginning of its impulse response and the "cutoff time"
* of the filter. The cutoff time is defined as the reciprocal of the
* lowpass-filter cut off frequence in Hz. For example, if the
* lowpass filter were a sinc function, Npc would be the index of the
* impulse-response lookup-table corresponding to the first zero-
* crossing of the sinc function. (The inverse first zero-crossing
* time of a sinc function equals its nominal cutoff frequency in Hz.)
* Npc must be a power of 2 due to the details of the current
* implementation. The default value of 512 is sufficiently high that
* using linear interpolation to fill in between the table entries
* gives approximately 16-bit accuracy in filter coefficients.
*
* Nhc - is log base 2 of Npc.
*
* Na - is the number of bits devoted to linear interpolation of the
* filter coefficients.
*
* Np - is Na + Nhc, the number of bits to the right of the binary point
* in the integer "time" variable. To the left of the point, it indexes
* the input array (X), and to the right, it is interpreted as a number
* between 0 and 1 sample of the input X. Np must be less than 16 in
* this implementation.
*
* Nh - is the number of bits in the filter coefficients. The sum of Nh and
* the number of bits in the input data (typically 16) cannot exceed 32.
* Thus Nh should be 16. The largest filter coefficient should nearly
* fill 16 bits (32767).
*
* Nb - is the number of bits in the input data. The sum of Nb and Nh cannot
* exceed 32.
*
* Nhxn - is the number of bits to right shift after multiplying each input
* sample times a filter coefficient. It can be as great as Nh and as
* small as 0. Nhxn = Nh-2 gives 2 guard bits in the multiply-add
* accumulation. If Nhxn=0, the accumulation will soon overflow 32 bits.
*
* Nhg - is the number of guard bits in mpy-add accumulation (equal to Nh-Nhxn)
*
* NLpScl - is the number of bits allocated to the unity-gain normalization
* factor. The output of the lowpass filter is multiplied by LpScl and
* then right-shifted NLpScl bits. To avoid overflow, we must have
* Nb+Nhg+NLpScl < 32.
*/
typedef char BOOL;
typedef short HWORD;
typedef unsigned short UHWORD;
typedef int WORD;
typedef unsigned int UWORD;
struct resample16_s;
typedef enum { RESAMPLE16_FAST, RESAMPLE16_SMALL, RESAMPLE16_LARGE, RESAMPLE_CUSTOM } resample16_filter_e;
WORD resample16(struct resample16_s *r, HWORD X[], int inCount, HWORD Y[]);
struct resample16_s* resample16_create(double factor, resample16_filter_e filter, BOOL interp);
void resample16_delete(struct resample16_s *r);
void resample16_flush(struct resample16_s *r);

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@@ -2,13 +2,14 @@
# "main" pseudo-component makefile.
#
# (Uses default behaviour of compiling all source files in directory, adding 'include' to include path.)
CFLAGS += -O3 -DPOSIX -DLINKALL -DLOOPBACK -DNO_FAAD -DRESAMPLE -DEMBEDDED -DTREMOR_ONLY -DBYTES_PER_FRAME=4 \
CFLAGS += -O3 -DPOSIX -DLINKALL -DLOOPBACK -DNO_FAAD -DRESAMPLE16 -DEMBEDDED -DTREMOR_ONLY -DBYTES_PER_FRAME=4 \
-I$(COMPONENT_PATH)/../components/codecs/inc \
-I$(COMPONENT_PATH)/../components/codecs/inc/mad \
-I$(COMPONENT_PATH)/../components/codecs/inc/alac \
-I$(COMPONENT_PATH)/../components/codecs/inc/helix-aac \
-I$(COMPONENT_PATH)/../components/codecs/inc/vorbis \
-I$(COMPONENT_PATH)/../components/codecs/inc/soxr \
-I$(COMPONENT_PATH)/../components/codecs/inc/resample16 \
-I$(COMPONENT_PATH)/../components/platform_esp32
LDFLAGS += -s

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@@ -111,6 +111,10 @@ static void usage(const char *argv0) {
" \t\t\t stopband_start = number in percent (Aliasing/imaging control. > passband_end),\n"
" \t\t\t phase_response = 0-100 (0 = minimum / 50 = linear / 100 = maximum)\n"
#endif
#if RESAMPLE16
" -R -u [params]\tResample, params = (i|m)[:i],\n"
" \t\t\t i = linear interpolation, m = 13 taps filter, i = interpolate filter coefficients\n"
#endif
#if DSD
#if ALSA
" -D [delay][:format]\tOutput device supports DSD, delay = optional delay switching between PCM and DSD in ms\n"
@@ -132,7 +136,7 @@ static void usage(const char *argv0) {
#if LINUX || FREEBSD || SUN
" -z \t\t\tDaemonize\n"
#endif
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
" -Z <rate>\t\tReport rate to server in helo as the maximum sample rate we can support\n"
#endif
" -t \t\t\tLicense terms\n"
@@ -183,6 +187,9 @@ static void usage(const char *argv0) {
#if RESAMPLE
" RESAMPLE"
#endif
#if RESAMPLE16
" RESAMPLE16"
#endif
#endif
#if FFMPEG
" FFMPEG"
@@ -339,7 +346,7 @@ int main(int argc, char **argv) {
* only allow '-Z <rate>' override of maxSampleRate
* reported by client if built with the capability to resample!
*/
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
"Z"
#endif
, opt) && optind < argc - 1) {
@@ -349,7 +356,7 @@ int main(int argc, char **argv) {
#if ALSA
"LX"
#endif
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
"uR"
#endif
#if DSD
@@ -531,7 +538,7 @@ int main(int argc, char **argv) {
exit(0);
break;
#endif
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
case 'u':
case 'R':
if (optind < argc && argv[optind] && argv[optind][0] != '-') {
@@ -783,7 +790,7 @@ else if(strstr(output_device,"DAC")!=NULL || strstr(output_device,"dac")!=NULL){
decode_init(log_decode, include_codecs, exclude_codecs);
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
if (resample) {
process_init(resample);
}

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@@ -39,7 +39,7 @@ extern struct codec *codec;
// macros to map to processing functions - currently only resample.c
// this can be made more generic when multiple processing mechanisms get added
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
#define SAMPLES_FUNC resample_samples
#define DRAIN_FUNC resample_drain
#define NEWSTREAM_FUNC resample_newstream

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@@ -323,8 +323,8 @@ bool resample_init(char *opt) {
r->scale = pow(10, -1.0 / 20);
// override recipe derived values with user specified values
r->q_precision = 0;
r->q_passband_end = 0;
r->q_stopband_begin = 0;
r->q_passband_end = 0.75;
r->q_stopband_begin = 1.25;
if (recipe && recipe[0] != '\0') {
if (strchr(recipe, 'v')) r->q_recipe = SOXR_VHQ;

163
main/resample16.c Normal file
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@@ -0,0 +1,163 @@
/*
* Squeezelite - lightweight headless squeezebox emulator
*
* (c) Adrian Smith 2012-2015, triode1@btinternet.com
* Ralph Irving 2015-2017, ralph_irving@hotmail.com
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
// upsampling using libsoxr - only included if RESAMPLE set
#include "squeezelite.h"
#if RESAMPLE16
#include <resample16.h>
extern log_level loglevel;
struct resample16 {
struct resample16_s *resampler;
bool max_rate;
bool exception;
bool interp;
resample16_filter_e filter;
};
static struct resample16 r;
void resample_samples(struct processstate *process) {
ssize_t odone;
odone = resample16(r.resampler, (HWORD*) process->inbuf, process->in_frames, (HWORD*) process->outbuf);
if (odone < 0) {
LOG_INFO("resample16 error");
return;
}
process->out_frames = odone;
process->total_in += process->in_frames;
process->total_out += odone;
}
bool resample_drain(struct processstate *process) {
process->out_frames = 0;
LOG_INFO("resample track complete");
resample16_delete(r.resampler);
r.resampler = NULL;
return true;
}
bool resample_newstream(struct processstate *process, unsigned raw_sample_rate, unsigned supported_rates[]) {
unsigned outrate = 0;
int i;
if (r.exception) {
// find direct match - avoid resampling
for (i = 0; supported_rates[i]; i++) {
if (raw_sample_rate == supported_rates[i]) {
outrate = raw_sample_rate;
break;
}
}
// else find next highest sync sample rate
while (!outrate && i >= 0) {
if (supported_rates[i] > raw_sample_rate && supported_rates[i] % raw_sample_rate == 0) {
outrate = supported_rates[i];
break;
}
i--;
}
}
if (!outrate) {
if (r.max_rate) {
// resample to max rate for device
outrate = supported_rates[0];
} else {
// resample to max sync sample rate
for (i = 0; supported_rates[i]; i++) {
if (supported_rates[i] % raw_sample_rate == 0 || raw_sample_rate % supported_rates[i] == 0) {
outrate = supported_rates[i];
break;
}
}
}
if (!outrate) {
outrate = supported_rates[0];
}
}
process->in_sample_rate = raw_sample_rate;
process->out_sample_rate = outrate;
if (r.resampler) {
resample16_delete(r.resampler);
r.resampler = NULL;
}
if (raw_sample_rate != outrate) {
LOG_INFO("resampling from %u -> %u", raw_sample_rate, outrate);
r.resampler = resample16_create((float) outrate / raw_sample_rate, RESAMPLE16_SMALL, false);
return true;
} else {
LOG_INFO("disable resampling - rates match");
return false;
}
}
void resample_flush(void) {
if (r.resampler) {
resample16_delete(r.resampler);
r.resampler = NULL;
}
}
bool resample_init(char *opt) {
char *filter = NULL, *interp = NULL;
r.resampler = NULL;
r.max_rate = false;
r.exception = false;
if (opt) {
filter = next_param(opt, ':');
interp = next_param(NULL, ':');
}
if (filter) {
if (*filter == 'm') r.filter = RESAMPLE16_SMALL;
else r.filter = RESAMPLE16_FAST;
}
if (interp && *interp == 'i') {
r.interp = true;
}
LOG_INFO("Resampling with filter %d %s", r.filter, r.interp ? "(interpolated)" : "");
return true;
}
#endif // #if RESAMPLE16

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@@ -132,6 +132,10 @@
#undef RESAMPLE
#define RESAMPLE 1 // resampling
#define PROCESS 1 // any sample processing (only resampling at present)
#elif defined(RESAMPLE16)
#undef RESAMPLE16
#define RESAMPLE16 1
#define PROCESS 1
#else
#define RESAMPLE 0
#define PROCESS 0
@@ -647,7 +651,7 @@ unsigned process_newstream(bool *direct, unsigned raw_sample_rate, unsigned supp
void process_init(char *opt);
#endif
#if RESAMPLE
#if RESAMPLE || RESAMPLE16
// resample.c
void resample_samples(struct processstate *process);
bool resample_drain(struct processstate *process);

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@@ -348,7 +348,7 @@ struct codec *register_vorbis(void) {
static struct codec ret = {
'o', // id
"ogg", // types
2048, // min read
4096, // min read
20480, // min space
vorbis_open, // open
vorbis_close, // close