/*
* Rate converter plugin using libavresample
* Copyright (c) 2014 by Anton Khirnov
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*/
#include <stdio.h>
#include <alsa/asoundlib.h>
#include <alsa/pcm_rate.h>
#include <libavresample/avresample.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
static unsigned int filter_size = 16;
static unsigned int phase_shift = 10; /* auto-adjusts */
static double cutoff = 0; /* auto-adjusts */
struct rate_src {
AVAudioResampleContext *avr;
unsigned int in_rate;
unsigned int out_rate;
unsigned int channels;
};
static snd_pcm_uframes_t input_frames(void *obj ATTRIBUTE_UNUSED,
snd_pcm_uframes_t frames)
{
return frames;
}
static snd_pcm_uframes_t output_frames(void *obj ATTRIBUTE_UNUSED,
snd_pcm_uframes_t frames)
{
return frames;
}
static void pcm_src_free(void *obj)
{
struct rate_src *rate = obj;
avresample_free(&rate->avr);
}
static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
{
struct rate_src *rate = obj;
int i, ir, or;
if (!rate->avr || rate->channels != info->channels) {
int ret;
pcm_src_free(rate);
rate->channels = info->channels;
ir = rate->in_rate = info->in.rate;
or = rate->out_rate = info->out.rate;
i = av_gcd(or, ir);
if (or > ir) {
phase_shift = or/i;
} else {
phase_shift = ir/i;
}
if (cutoff <= 0.0) {
cutoff = 1.0 - 1.0/filter_size;
if (cutoff < 0.80)
cutoff = 0.80;
}
rate->avr = avresample_alloc_context();
if (!rate->avr)
return -ENOMEM;
av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0);
av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0);
av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0);
av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0);
av_opt_set_int(rate->avr, "filter_size", filter_size, 0);
av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0);
av_opt_set_double(rate->avr, "cutoff", cutoff, 0);
ret = avresample_open(rate->avr);
if (ret < 0) {
avresample_free(&rate->avr);
return -EINVAL;
}
}
return 0;
}
static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
{
struct rate_src *rate = obj;
if (info->out.rate != rate->out_rate || info->in.rate != rate->in_rate)
pcm_src_init(obj, info);
return 0;
}
static void pcm_src_reset(void *obj)
{
struct rate_src *rate = obj;
if (rate->avr) {
#if 0
avresample_close(rate->avr);
avresample_open(rate->avr);
#endif
}
}
static void pcm_src_convert_s16(void *obj, int16_t *dst,
unsigned int dst_frames,
const int16_t *src,
unsigned int src_frames)
{
struct rate_src *rate = obj;
int chans = rate->channels;
unsigned int total_in = avresample_get_delay(rate->avr) + src_frames;
avresample_convert(rate->avr, (uint8_t **)&dst, dst_frames * chans * 2, dst_frames,
(uint8_t **)&src, src_frames * chans * 2, src_frames);
avresample_set_compensation(rate->avr,
total_in - src_frames > filter_size ? 0 : 1, src_frames);
}
static void pcm_src_close(void *obj)
{
pcm_src_free(obj);
}
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
static int get_supported_rates(void *obj ATTRIBUTE_UNUSED,
unsigned int *rate_min,
unsigned int *rate_max)
{
*rate_min = *rate_max = 0; /* both unlimited */
return 0;
}
static void dump(void *obj ATTRIBUTE_UNUSED, snd_output_t *out)
{
snd_output_printf(out, "Converter: libavr\n");
}
#endif
static snd_pcm_rate_ops_t pcm_src_ops = {
.close = pcm_src_close,
.init = pcm_src_init,
.free = pcm_src_free,
.reset = pcm_src_reset,
.adjust_pitch = pcm_src_adjust_pitch,
.convert_s16 = pcm_src_convert_s16,
.input_frames = input_frames,
.output_frames = output_frames,
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
.version = SND_PCM_RATE_PLUGIN_VERSION,
.get_supported_rates = get_supported_rates,
.dump = dump,
#endif
};
int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)
{
struct rate_src *rate;
#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002
if (version != SND_PCM_RATE_PLUGIN_VERSION) {
fprintf(stderr, "Invalid rate plugin version %x\n", version);
return -EINVAL;
}
#endif
rate = calloc(1, sizeof(*rate));
if (!rate)
return -ENOMEM;
*objp = rate;
rate->avr = NULL;
#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
if (version == 0x010001)
memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t));
else
#endif
*ops = pcm_src_ops;
return 0;
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavrate)(unsigned int version, void **objp,
snd_pcm_rate_ops_t *ops)
{
return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavrate_higher)(unsigned int version,
void **objp, snd_pcm_rate_ops_t *ops)
{
filter_size = 64;
return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavrate_high)(unsigned int version,
void **objp, snd_pcm_rate_ops_t *ops)
{
filter_size = 32;
return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavrate_fast)(unsigned int version,
void **objp, snd_pcm_rate_ops_t *ops)
{
filter_size = 8;
return pcm_src_open(version, objp, ops);
}
int SND_PCM_RATE_PLUGIN_ENTRY(lavrate_faster)(unsigned int version,
void **objp, snd_pcm_rate_ops_t *ops)
{
filter_size = 4;
return pcm_src_open(version, objp, ops);
}