Dne 5.4.2018 v 05:05 Flavio Protasio Ribeiro napsal(a):
Hi Folks,
The libsamplerate and speex rate converter plugins only implement the
convert_s16 callback from snd_pcm_rate_ops_t. This has ALSA convert
the buffer to int16_t before handing it over to the plugin for
resampling. On devices with more than 16 effective bits, this implies
words gets truncated and bits are lost.
Specifically for libsamplerate, the resampling is internally
implemented with floats, so truncating to 16 bits does not offer a
computational benefit.
I get why only convert_s16 is implemented for plugins that aren't
part of the main alsa-lib package. The more generic convert callback
comes with the burden having the plugin support conversion from/to
any of the ALSA data types (or alternatively, having the plugin
source include plugin_ops.h and its dependencies). So here's what I
propose: replace convert_s16 with a convert_s32 callback which uses
int32_t instead of int16_t. This would preserve all bits from the
device and keep the plugin interface simple. Of course this means
plugins have to be updated to use convert_s32 and rebuilt.
Hi,
Many years ago I tried to extend the rate API to float, the native
formats of libsamplerate and speex.
I go stuck on float support of all platforms, perhaps you can get
something from the discussion and patches.
http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/041059.html
http://mailman.alsa-project.org/pipermail/alsa-devel/2011-July/041503.html
If we succeed in adding >16 bit API to the rate plugin, perhaps
supporting libsoxr would be the next step which would bring great value
(performance vs. CPU load) to alsa-lib. Unfortunately that is not just
about conversion, there are other quirks to iron out...
Anyway, thanks a lot and good luck to your very useful endeavour.
Best regards,
Pavel.
>From cf3b349ea2a2f55e9cfdece3d8123edc3d620719 Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@xxxxxxxxxxx>
Date: Tue, 28 Jun 2011 08:50:23 +0200
Subject: [PATCH - rate plugin 1/1] Support for float samples in rate converters
Some rate converters have native float resolution, no need
to loose information by converting to s16.
Signed-off-by: Pavel Hofman <pavel.hofman@xxxxxxxxxxx>
diff --git a/include/pcm_rate.h b/include/pcm_rate.h
index 4d70df2..e92900b 100644
--- a/include/pcm_rate.h
+++ b/include/pcm_rate.h
@@ -91,6 +91,11 @@ typedef struct snd_pcm_rate_ops {
void (*convert_s16)(void *obj, int16_t *dst, unsigned int dst_frames,
const int16_t *src, unsigned int src_frames);
/**
+ * convert a float interleaved-data array; exclusive with convert
+ */
+ void (*convert_float)(void *obj, float *dst, unsigned int dst_frames,
+ const float *src, unsigned int src_frames);
+ /**
* compute the frame size for input
*/
snd_pcm_uframes_t (*input_frames)(void *obj, snd_pcm_uframes_t frames);
diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c
index 70e30e5..9f4ec23 100644
--- a/src/pcm/pcm_rate.c
+++ b/src/pcm/pcm_rate.c
@@ -35,7 +35,7 @@
#include "iatomic.h"
#include "plugin_ops.h"
-
+typedef float float_t;
#if 0
#define DEBUG_REFINE
#endif
@@ -66,8 +66,8 @@ struct _snd_pcm_rate {
snd_pcm_rate_ops_t ops;
unsigned int get_idx;
unsigned int put_idx;
- int16_t *src_buf;
- int16_t *dst_buf;
+ void *src_buf;
+ void *dst_buf;
int start_pending; /* start is triggered but not commited to slave */
snd_htimestamp_t trigger_tstamp;
unsigned int plugin_version;
@@ -310,13 +310,23 @@ static int snd_pcm_rate_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params)
rate->sareas[chn].step = swidth;
}
- if (rate->ops.convert_s16) {
+ if (rate->ops.convert_float) {
+ rate->get_idx = snd_pcm_linear_get_index(rate->info.in.format, SND_PCM_FORMAT_S32);
+ rate->put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S32, rate->info.out.format);
+ free(rate->src_buf);
+ rate->src_buf = (float_t *) malloc(channels * rate->info.in.period_size * sizeof(float_t));
+ free(rate->dst_buf);
+ rate->dst_buf = (float_t *) malloc(channels * rate->info.out.period_size * sizeof(float_t));
+ if (! rate->src_buf || ! rate->dst_buf)
+ goto error;
+ }
+ else if (rate->ops.convert_s16) {
rate->get_idx = snd_pcm_linear_get_index(rate->info.in.format, SND_PCM_FORMAT_S16);
rate->put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, rate->info.out.format);
free(rate->src_buf);
- rate->src_buf = malloc(channels * rate->info.in.period_size * 2);
+ rate->src_buf = (int16_t *) malloc(channels * rate->info.in.period_size * 2);
free(rate->dst_buf);
- rate->dst_buf = malloc(channels * rate->info.out.period_size * 2);
+ rate->dst_buf = (int16_t *) malloc(channels * rate->info.out.period_size * 2);
if (! rate->src_buf || ! rate->dst_buf)
goto error;
}
@@ -503,6 +513,85 @@ static void convert_from_s16(snd_pcm_rate_t *rate, const int16_t *buf,
}
}
}
+static void convert_to_float(snd_pcm_rate_t *rate, float_t *buf,
+ const snd_pcm_channel_area_t *areas,
+ snd_pcm_uframes_t offset, unsigned int frames,
+ unsigned int channels)
+{
+#ifndef DOC_HIDDEN
+#define GET32_LABELS
+#include "plugin_ops.h"
+#undef GET32_LABELS
+#endif /* DOC_HIDDEN */
+ void *get32 = get32_labels[rate->get_idx];
+ const char *src;
+ int32_t sample;
+ const char *srcs[channels];
+ int src_step[channels];
+ unsigned int c;
+
+ for (c = 0; c < channels; c++) {
+ srcs[c] = snd_pcm_channel_area_addr(areas + c, offset);
+ src_step[c] = snd_pcm_channel_area_step(areas + c);
+ }
+
+ while (frames--) {
+ for (c = 0; c < channels; c++) {
+ src = srcs[c];
+ // converting the source format to int32 first
+ goto *get32;
+#ifndef DOC_HIDDEN
+#define GET32_END after_get
+#include "plugin_ops.h"
+#undef GET32_END
+#endif /* DOC_HIDDEN */
+ after_get:
+ // converting to float for the plugin
+ *buf++ = (float_t) sample;
+ srcs[c] += src_step[c];
+ }
+ }
+}
+
+static void convert_from_float(snd_pcm_rate_t *rate, const float_t *buf,
+ const snd_pcm_channel_area_t *areas,
+ snd_pcm_uframes_t offset, unsigned int frames,
+ unsigned int channels)
+{
+#ifndef DOC_HIDDEN
+#define PUT32_LABELS
+#include "plugin_ops.h"
+#undef PUT32_LABELS
+#endif /* DOC_HIDDEN */
+ void *put32 = put32_labels[rate->put_idx];
+ char *dst;
+ int32_t sample;
+ char *dsts[channels];
+ int dst_step[channels];
+ unsigned int c;
+
+ for (c = 0; c < channels; c++) {
+ dsts[c] = snd_pcm_channel_area_addr(areas + c, offset);
+ dst_step[c] = snd_pcm_channel_area_step(areas + c);
+ }
+
+ while (frames--) {
+ for (c = 0; c < channels; c++) {
+ dst = dsts[c];
+ // plugin returned float, converting to int32 first
+ sample = (int32_t) *buf++;
+ // converting int32 to the destination format
+ goto *put32;
+#ifndef DOC_HIDDEN
+#define PUT32_END after_put
+#include "plugin_ops.h"
+#undef PUT32_END
+#endif /* DOC_HIDDEN */
+ after_put:
+ dsts[c] += dst_step[c];
+ }
+ }
+}
static void do_convert(const snd_pcm_channel_area_t *dst_areas,
snd_pcm_uframes_t dst_offset, unsigned int dst_frames,
@@ -511,18 +600,36 @@ static void do_convert(const snd_pcm_channel_area_t *dst_areas,
unsigned int channels,
snd_pcm_rate_t *rate)
{
- if (rate->ops.convert_s16) {
+ if (rate->ops.convert_float) {
+ const float_t *src;
+ float_t *dst;
+ if (! rate->src_buf)
+ src = src_areas->addr + src_offset * sizeof(float_t) * channels;
+ else {
+ convert_to_float(rate, rate->src_buf, src_areas, src_offset,
+ src_frames, channels);
+ src = rate->src_buf;
+ }
+ if (! rate->dst_buf)
+ dst = dst_areas->addr + dst_offset * sizeof(float_t) * channels;
+ else
+ dst = rate->dst_buf;
+ rate->ops.convert_float(rate->obj, dst, dst_frames, src, src_frames);
+ if (dst == rate->dst_buf)
+ convert_from_float(rate, rate->dst_buf, dst_areas, dst_offset,
+ dst_frames, channels);
+ } else if (rate->ops.convert_s16) {
const int16_t *src;
int16_t *dst;
if (! rate->src_buf)
- src = src_areas->addr + src_offset * 2 * channels;
+ src = src_areas->addr + src_offset * sizeof(int16_t) * channels;
else {
convert_to_s16(rate, rate->src_buf, src_areas, src_offset,
src_frames, channels);
src = rate->src_buf;
}
if (! rate->dst_buf)
- dst = dst_areas->addr + dst_offset * 2 * channels;
+ dst = dst_areas->addr + dst_offset * sizeof(int16_t) * channels;
else
dst = rate->dst_buf;
rate->ops.convert_s16(rate->obj, dst, dst_frames, src, src_frames);
@@ -1409,7 +1516,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name,
}
#endif
- if (! rate->ops.init || ! (rate->ops.convert || rate->ops.convert_s16) ||
+ if (! rate->ops.init || ! (rate->ops.convert || rate->ops.convert_s16 || rate->ops.convert_float) ||
! rate->ops.input_frames || ! rate->ops.output_frames) {
SNDERR("Inproper rate plugin %s initialization", type);
snd_pcm_close(pcm);
--
1.7.0.4
>From 34fb26dc0a95f7d1950abe2582cf79219d1876a9 Mon Sep 17 00:00:00 2001
From: Pavel Hofman <pavel.hofman@xxxxxxxxxxx>
Date: Tue, 28 Jun 2011 08:55:08 +0200
Subject: [PATCH - resample 1/1] Using float instead of s16 for libsamplerate
Natively libsamplerate operates in floats. Using the new convert_float API instead of dropping resoluton in convert_s16.
Signed-off-by: Pavel Hofman <pavel.hofman@xxxxxxxxxxx>
diff --git a/rate/rate_samplerate.c b/rate/rate_samplerate.c
index 53a0627..c612983 100644
--- a/rate/rate_samplerate.c
+++ b/rate/rate_samplerate.c
@@ -23,12 +23,12 @@
#include <alsa/asoundlib.h>
#include <alsa/pcm_rate.h>
+typedef float float_t;
+
struct rate_src {
double ratio;
int converter;
unsigned int channels;
- float *src_buf;
- float *dst_buf;
SRC_STATE *state;
SRC_DATA data;
};
@@ -53,9 +53,6 @@ static void pcm_src_free(void *obj)
{
struct rate_src *rate = obj;
- free(rate->src_buf);
- free(rate->dst_buf);
- rate->src_buf = rate->dst_buf = NULL;
if (rate->state) {
src_delete(rate->state);
@@ -79,17 +76,6 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
rate->ratio = (double)info->out.rate / (double)info->in.rate;
- free(rate->src_buf);
- rate->src_buf = malloc(sizeof(float) * rate->channels * info->in.period_size);
- free(rate->dst_buf);
- rate->dst_buf = malloc(sizeof(float) * rate->channels * info->out.period_size);
- if (! rate->src_buf || ! rate->dst_buf) {
- pcm_src_free(rate);
- return -ENOMEM;
- }
-
- rate->data.data_in = rate->src_buf;
- rate->data.data_out = rate->dst_buf;
rate->data.src_ratio = rate->ratio;
rate->data.end_of_input = 0;
@@ -112,8 +98,8 @@ static void pcm_src_reset(void *obj)
src_reset(rate->state);
}
-static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames,
- const int16_t *src, unsigned int src_frames)
+static void pcm_src_convert_float(void *obj, float_t *dst, unsigned int dst_frames,
+ const float_t *src, unsigned int src_frames)
{
struct rate_src *rate = obj;
unsigned int ofs;
@@ -121,15 +107,17 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames
rate->data.input_frames = src_frames;
rate->data.output_frames = dst_frames;
rate->data.end_of_input = 0;
+ rate->data.data_in = src;
+ rate->data.data_out = dst;
+
- src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels);
src_process(rate->state, &rate->data);
if (rate->data.output_frames_gen < dst_frames)
ofs = dst_frames - rate->data.output_frames_gen;
else
ofs = 0;
- src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels,
- rate->data.output_frames_gen * rate->channels);
+ /*src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels,
+ rate->data.output_frames_gen * rate->channels); */
}
static void pcm_src_close(void *obj)
@@ -157,7 +145,7 @@ static snd_pcm_rate_ops_t pcm_src_ops = {
.free = pcm_src_free,
.reset = pcm_src_reset,
.adjust_pitch = pcm_src_adjust_pitch,
- .convert_s16 = pcm_src_convert_s16,
+ .convert_float = pcm_src_convert_float,
.input_frames = input_frames,
.output_frames = output_frames,
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
--
1.7.0.4
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