Commit 5ba90cd8 authored by Max Kellermann's avatar Max Kellermann

pcm/PcmResampler: convert to abstract interface

The PcmResampler interface is implemented by the two classes FallbackPcmResampler and LibsampleratePcmResampler. This prepares for adding more resampler libraries.
parent e9127523
......@@ -339,10 +339,10 @@ libpcm_a_SOURCES = \
src/pcm/PcmFormat.cxx src/pcm/PcmFormat.hxx \
src/pcm/FormatConverter.cxx src/pcm/FormatConverter.hxx \
src/pcm/ChannelsConverter.cxx src/pcm/ChannelsConverter.hxx \
src/pcm/Resampler.hxx \
src/pcm/GlueResampler.cxx src/pcm/GlueResampler.hxx \
src/pcm/PcmResample.cxx src/pcm/PcmResample.hxx \
src/pcm/PcmResampleFallback.cxx \
src/pcm/PcmResampleInternal.hxx \
src/pcm/FallbackResampler.cxx src/pcm/FallbackResampler.hxx \
src/pcm/ConfiguredResampler.cxx src/pcm/ConfiguredResampler.hxx \
src/pcm/PcmDither.cxx src/pcm/PcmDither.hxx \
src/pcm/PcmPrng.hxx \
src/pcm/PcmUtils.hxx
......@@ -354,7 +354,8 @@ PCM_LIBS = \
$(SAMPLERATE_LIBS)
if HAVE_LIBSAMPLERATE
libpcm_a_SOURCES += src/pcm/PcmResampleLibsamplerate.cxx
libpcm_a_SOURCES += \
src/pcm/LibsamplerateResampler.cxx src/pcm/LibsamplerateResampler.hxx
endif
# File system library
......
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "ConfiguredResampler.hxx"
#include "FallbackResampler.hxx"
#ifdef HAVE_LIBSAMPLERATE
#include "LibsamplerateResampler.hxx"
#include "ConfigGlobal.hxx"
#include "ConfigOption.hxx"
#endif
#include <string.h>
#ifdef HAVE_LIBSAMPLERATE
static bool lsr_enabled;
#endif
bool
pcm_resampler_global_init(Error &error)
{
#ifdef HAVE_LIBSAMPLERATE
const char *converter =
config_get_string(CONF_SAMPLERATE_CONVERTER, "");
lsr_enabled = strcmp(converter, "internal") != 0;
if (lsr_enabled)
return pcm_resample_lsr_global_init(converter, error);
else
return true;
#else
(void)error;
return true;
#endif
}
PcmResampler *
pcm_resampler_create()
{
#ifdef HAVE_LIBSAMPLERATE
if (lsr_enabled)
return new LibsampleratePcmResampler();
#endif
return new FallbackPcmResampler();
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_CONFIGURED_RESAMPLER_HXX
#define MPD_CONFIGURED_RESAMPLER_HXX
#include "check.h"
class Error;
class PcmResampler;
bool
pcm_resampler_global_init(Error &error);
/**
* Create a #PcmResampler instance from the implementation class
* configured in mpd.conf.
*/
PcmResampler *
pcm_resampler_create();
#endif
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "FallbackResampler.hxx"
#include <assert.h>
AudioFormat
FallbackPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate,
gcc_unused Error &error)
{
assert(af.IsValid());
assert(audio_valid_sample_rate(new_sample_rate));
switch (af.format) {
case SampleFormat::UNDEFINED:
assert(false);
gcc_unreachable();
case SampleFormat::S8:
af.format = SampleFormat::S16;
break;
case SampleFormat::S16:
case SampleFormat::FLOAT:
case SampleFormat::S24_P32:
case SampleFormat::S32:
break;
case SampleFormat::DSD:
af.format = SampleFormat::FLOAT;
break;
}
format = af;
out_rate = new_sample_rate;
AudioFormat result = af;
result.sample_rate = new_sample_rate;
return result;
}
void
FallbackPcmResampler::Close()
{
}
template<typename T>
static ConstBuffer<T>
pcm_resample_fallback(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
ConstBuffer<T> src,
unsigned dest_rate)
{
unsigned dest_pos = 0;
unsigned src_frames = src.size / channels;
unsigned dest_frames =
(src_frames * dest_rate + src_rate - 1) / src_rate;
unsigned dest_samples = dest_frames * channels;
size_t dest_size = dest_samples * sizeof(*src.data);
T *dest_buffer = (T *)buffer.Get(dest_size);
assert((src.size % channels) == 0);
switch (channels) {
case 1:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
dest_buffer[dest_pos++] = src.data[src_pos];
}
break;
case 2:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
src_pos &= ~1;
dest_buffer[dest_pos++] = src.data[src_pos];
dest_buffer[dest_pos++] = src.data[src_pos + 1];
}
break;
}
return { dest_buffer, dest_samples };
}
template<typename T>
static ConstBuffer<void>
pcm_resample_fallback_void(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
ConstBuffer<void> src,
unsigned dest_rate)
{
const auto typed_src = ConstBuffer<T>::FromVoid(src);
return pcm_resample_fallback(buffer, channels, src_rate, typed_src,
dest_rate);
}
ConstBuffer<void>
FallbackPcmResampler::Resample(ConstBuffer<void> src, gcc_unused Error &error)
{
switch (format.format) {
case SampleFormat::UNDEFINED:
case SampleFormat::S8:
case SampleFormat::DSD:
assert(false);
gcc_unreachable();
case SampleFormat::S16:
return pcm_resample_fallback_void<int16_t>(buffer,
format.channels,
format.sample_rate,
src,
out_rate);
case SampleFormat::FLOAT:
case SampleFormat::S24_P32:
case SampleFormat::S32:
return pcm_resample_fallback_void<int32_t>(buffer,
format.channels,
format.sample_rate,
src,
out_rate);
}
assert(false);
gcc_unreachable();
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_PCM_FALLBACK_RESAMPLER_HXX
#define MPD_PCM_FALLBACK_RESAMPLER_HXX
#include "Resampler.hxx"
#include "PcmBuffer.hxx"
#include "AudioFormat.hxx"
/**
* A naive resampler that is used when no external library was found
* (or when the user explicitly asks for bad quality).
*/
class FallbackPcmResampler final : public PcmResampler {
AudioFormat format;
unsigned out_rate;
PcmBuffer buffer;
public:
virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
Error &error) override;
virtual void Close() override;
virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
Error &error) override;
};
#endif
......@@ -19,76 +19,67 @@
#include "config.h"
#include "GlueResampler.hxx"
#include "PcmConvert.hxx"
#include "PcmFormat.hxx"
#include "util/ConstBuffer.hxx"
#include "util/Error.hxx"
#include "ConfiguredResampler.hxx"
#include "Resampler.hxx"
#include <assert.h>
GluePcmResampler::GluePcmResampler()
:resampler(pcm_resampler_create()) {}
GluePcmResampler::~GluePcmResampler()
{
delete resampler;
}
bool
GluePcmResampler::Open(AudioFormat _src_format, unsigned _new_sample_rate,
gcc_unused Error &error)
GluePcmResampler::Open(AudioFormat src_format, unsigned new_sample_rate,
Error &error)
{
src_format = _src_format;
new_sample_rate = _new_sample_rate;
assert(src_format.IsValid());
assert(audio_valid_sample_rate(new_sample_rate));
AudioFormat requested_format = src_format;
AudioFormat dest_format = resampler->Open(requested_format,
new_sample_rate,
error);
if (!dest_format.IsValid())
return false;
assert(requested_format.channels == src_format.channels);
assert(dest_format.channels == src_format.channels);
assert(dest_format.sample_rate == new_sample_rate);
if (requested_format.format != src_format.format &&
!format_converter.Open(src_format.format, requested_format.format,
error))
return false;
src_sample_format = src_format.format;
requested_sample_format = requested_format.format;
output_sample_format = dest_format.format;
return true;
}
void
GluePcmResampler::Close()
{
resampler.Reset();
if (requested_sample_format != src_sample_format)
format_converter.Close();
resampler->Close();
}
ConstBuffer<void>
GluePcmResampler::Resample(ConstBuffer<void> src, Error &error)
{
const void *result;
size_t size;
switch (src_format.format) {
case SampleFormat::S16:
result = resampler.Resample16(src_format.channels,
src_format.sample_rate,
(const int16_t *)src.data,
src.size,
new_sample_rate, &size,
error);
break;
case SampleFormat::S24_P32:
result = resampler.Resample24(src_format.channels,
src_format.sample_rate,
(const int32_t *)src.data,
src.size,
new_sample_rate, &size,
error);
break;
case SampleFormat::S32:
result = resampler.Resample24(src_format.channels,
src_format.sample_rate,
(const int32_t *)src.data,
src.size,
new_sample_rate, &size,
error);
break;
case SampleFormat::FLOAT:
result = resampler.ResampleFloat(src_format.channels,
src_format.sample_rate,
(const float *)src.data,
src.size,
new_sample_rate, &size,
error);
break;
assert(!src.IsNull());
default:
error.Format(pcm_convert_domain,
"Resampling %s is not implemented",
sample_format_to_string(src_format.format));
return nullptr;
if (requested_sample_format != src_sample_format) {
src = format_converter.Convert(src, error);
if (src.IsNull())
return nullptr;
}
return { result, size };
return resampler->Resample(src, error);
}
......@@ -22,22 +22,41 @@
#include "check.h"
#include "AudioFormat.hxx"
#include "PcmResample.hxx"
#include "FormatConverter.hxx"
class Error;
class PcmResampler;
template<typename T> struct ConstBuffer;
/**
* A glue class that integrates a #PcmResampler and automatically
* converts source data to the sample format required by the
* #PcmResampler instance.
*/
class GluePcmResampler {
PcmResampler resampler;
PcmResampler *const resampler;
SampleFormat src_sample_format, requested_sample_format;
SampleFormat output_sample_format;
AudioFormat src_format;
unsigned new_sample_rate;
/**
* This object converts input data to the sample format
* requested by the #PcmResampler.
*/
PcmFormatConverter format_converter;
public:
GluePcmResampler();
~GluePcmResampler();
bool Open(AudioFormat src_format, unsigned new_sample_rate,
Error &error);
void Close();
SampleFormat GetOutputSampleFormat() const {
return output_sample_format;
}
ConstBuffer<void> Resample(ConstBuffer<void> src, Error &error);
};
......
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "LibsamplerateResampler.hxx"
#include "util/ASCII.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "Log.hxx"
#include <assert.h>
#include <stdlib.h>
#include <string.h>
static constexpr Domain libsamplerate_domain("libsamplerate");
static int lsr_converter = SRC_SINC_FASTEST;
static bool
lsr_parse_converter(const char *s)
{
assert(s != nullptr);
if (*s == 0)
return true;
char *endptr;
long l = strtol(s, &endptr, 10);
if (*endptr == 0 && src_get_name(l) != nullptr) {
lsr_converter = l;
return true;
}
size_t length = strlen(s);
for (int i = 0;; ++i) {
const char *name = src_get_name(i);
if (name == nullptr)
break;
if (StringEqualsCaseASCII(s, name, length)) {
lsr_converter = i;
return true;
}
}
return false;
}
bool
pcm_resample_lsr_global_init(const char *converter, Error &error)
{
if (!lsr_parse_converter(converter)) {
error.Format(libsamplerate_domain,
"unknown samplerate converter '%s'", converter);
return false;
}
FormatDebug(libsamplerate_domain,
"libsamplerate converter '%s'",
src_get_name(lsr_converter));
return true;
}
AudioFormat
LibsampleratePcmResampler::Open(AudioFormat &af, unsigned new_sample_rate,
Error &error)
{
assert(af.IsValid());
assert(audio_valid_sample_rate(new_sample_rate));
src_rate = af.sample_rate;
dest_rate = new_sample_rate;
channels = af.channels;
/* libsamplerate works with floating point samples */
af.format = SampleFormat::FLOAT;
int src_error;
state = src_new(lsr_converter, channels, &src_error);
if (!state) {
error.Format(libsamplerate_domain, src_error,
"libsamplerate initialization has failed: %s",
src_strerror(src_error));
return AudioFormat::Undefined();
}
memset(&data, 0, sizeof(data));
data.src_ratio = double(new_sample_rate) / double(af.sample_rate);
FormatDebug(libsamplerate_domain,
"setting samplerate conversion ratio to %.2lf",
data.src_ratio);
src_set_ratio(state, data.src_ratio);
AudioFormat result = af;
result.sample_rate = new_sample_rate;
return result;
}
void
LibsampleratePcmResampler::Close()
{
state = src_delete(state);
}
static bool
src_process(SRC_STATE *state, SRC_DATA *data, Error &error)
{
int result = src_process(state, data);
if (result != 0) {
error.Format(libsamplerate_domain, result,
"libsamplerate has failed: %s",
src_strerror(result));
return false;
}
return true;
}
inline ConstBuffer<float>
LibsampleratePcmResampler::Resample2(ConstBuffer<float> src, Error &error)
{
assert(src.size % channels == 0);
const unsigned src_frames = src.size / channels;
const unsigned dest_frames =
(src_frames * dest_rate + src_rate - 1) / src_rate;
size_t data_out_size = dest_frames * sizeof(float) * channels;
data.data_in = const_cast<float *>(src.data);
data.data_out = (float *)buffer.Get(data_out_size);
data.input_frames = src_frames;
data.output_frames = dest_frames;
if (!src_process(state, &data, error))
return nullptr;
return ConstBuffer<float>(data.data_out,
data.output_frames_gen * channels);
}
ConstBuffer<void>
LibsampleratePcmResampler::Resample(ConstBuffer<void> src, Error &error)
{
return Resample2(ConstBuffer<float>::FromVoid(src), error).ToVoid();
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_PCM_LIBSAMPLERATE_RESAMPLER_HXX
#define MPD_PCM_LIBSAMPLERATE_RESAMPLER_HXX
#include "Resampler.hxx"
#include "PcmBuffer.hxx"
#include "AudioFormat.hxx"
#include <samplerate.h>
/**
* A resampler using libsamplerate.
*/
class LibsampleratePcmResampler final : public PcmResampler {
unsigned src_rate, dest_rate;
unsigned channels;
SRC_STATE *state;
SRC_DATA data;
PcmBuffer buffer;
public:
virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
Error &error) override;
virtual void Close() override;
virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
Error &error) override;
private:
ConstBuffer<float> Resample2(ConstBuffer<float> src, Error &error);
};
bool
pcm_resample_lsr_global_init(const char *converter, Error &error);
#endif
......@@ -19,6 +19,7 @@
#include "config.h"
#include "PcmConvert.hxx"
#include "ConfiguredResampler.hxx"
#include "AudioFormat.hxx"
#include "util/ConstBuffer.hxx"
#include "util/Error.hxx"
......@@ -33,7 +34,7 @@ const Domain pcm_convert_domain("pcm_convert");
bool
pcm_convert_global_init(Error &error)
{
return pcm_resample_global_init(error);
return pcm_resampler_global_init(error);
}
PcmConvert::PcmConvert()
......@@ -66,39 +67,51 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
if (format.format == SampleFormat::DSD)
format.format = SampleFormat::FLOAT;
if (format.format != dest_format.format &&
!format_converter.Open(format.format, dest_format.format, error))
enable_resampler = format.sample_rate != dest_format.sample_rate;
if (enable_resampler) {
if (!resampler.Open(format, dest_format.sample_rate, error))
return false;
format.format = resampler.GetOutputSampleFormat();
format.sample_rate = dest_format.sample_rate;
}
enable_format = format.format != dest_format.format;
if (enable_format &&
!format_converter.Open(format.format, dest_format.format, error)) {
if (enable_resampler)
resampler.Close();
return false;
}
format.format = dest_format.format;
if (format.channels != dest_format.channels &&
enable_channels = format.channels != dest_format.channels;
if (enable_channels &&
!channels_converter.Open(format.format, format.channels,
dest_format.channels, error)) {
format_converter.Close();
if (enable_format)
format_converter.Close();
if (enable_resampler)
resampler.Close();
return false;
}
if (format.sample_rate != dest_format.sample_rate &&
!resampler.Open(format, dest_format.sample_rate, error))
return false;
return true;
}
void
PcmConvert::Close()
{
if (src_format.channels != dest_format.channels)
if (enable_channels)
channels_converter.Close();
if (src_format.format != dest_format.format)
if (enable_format)
format_converter.Close();
if (enable_resampler)
resampler.Close();
dsd.Reset();
if (src_format.sample_rate != dest_format.sample_rate)
resampler.Close();
#ifndef NDEBUG
src_format.Clear();
dest_format.Clear();
......@@ -127,28 +140,29 @@ PcmConvert::Convert(const void *src, size_t src_size,
format.format = SampleFormat::FLOAT;
}
if (format.format != dest_format.format) {
buffer = format_converter.Convert(buffer, error);
if (enable_resampler) {
buffer = resampler.Resample(buffer, error);
if (buffer.IsNull())
return nullptr;
format.format = dest_format.format;
format.format = resampler.GetOutputSampleFormat();
format.sample_rate = dest_format.sample_rate;
}
if (format.channels != dest_format.channels) {
buffer = channels_converter.Convert(buffer, error);
if (enable_format) {
buffer = format_converter.Convert(buffer, error);
if (buffer.IsNull())
return nullptr;
format.channels = dest_format.channels;
format.format = dest_format.format;
}
if (format.sample_rate != dest_format.sample_rate) {
buffer = resampler.Resample(buffer, error);
if (enable_channels) {
buffer = channels_converter.Convert(buffer, error);
if (buffer.IsNull())
return nullptr;
format.sample_rate = dest_format.sample_rate;
format.channels = dest_format.channels;
}
*dest_size_r = buffer.size;
......
......@@ -41,12 +41,14 @@ class Domain;
class PcmConvert {
PcmDsd dsd;
GluePcmResampler resampler;
PcmFormatConverter format_converter;
PcmChannelsConverter channels_converter;
GluePcmResampler resampler;
AudioFormat src_format, dest_format;
bool enable_resampler, enable_format, enable_channels;
public:
PcmConvert();
~PcmConvert();
......
......@@ -442,7 +442,6 @@ ConvertToFloat(float *dest,
constexpr float factor = 0.5 / (1 << (Traits::BITS - 2));
while (src != end)
*dest++ = float(*src++) * factor;
}
template<SampleFormat F, class Traits=SampleTraits<F>>
......
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "PcmResampleInternal.hxx"
#ifdef HAVE_LIBSAMPLERATE
#include "ConfigGlobal.hxx"
#include "ConfigOption.hxx"
#endif
#include <string.h>
#ifdef HAVE_LIBSAMPLERATE
static bool lsr_enabled;
#endif
#ifdef HAVE_LIBSAMPLERATE
static bool
pcm_resample_lsr_enabled(void)
{
return lsr_enabled;
}
#endif
bool
pcm_resample_global_init(Error &error)
{
#ifdef HAVE_LIBSAMPLERATE
const char *converter =
config_get_string(CONF_SAMPLERATE_CONVERTER, "");
lsr_enabled = strcmp(converter, "internal") != 0;
if (lsr_enabled)
return pcm_resample_lsr_global_init(converter, error);
else
return true;
#else
(void)error;
return true;
#endif
}
PcmResampler::PcmResampler()
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
pcm_resample_lsr_init(this);
#endif
}
PcmResampler::~PcmResampler()
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
pcm_resample_lsr_deinit(this);
#endif
}
void
PcmResampler::Reset()
{
#ifdef HAVE_LIBSAMPLERATE
pcm_resample_lsr_reset(this);
#endif
}
const float *
PcmResampler::ResampleFloat(unsigned channels, unsigned src_rate,
const float *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r)
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
return pcm_resample_lsr_float(this, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r,
error_r);
#else
(void)error_r;
#endif
/* sizeof(float)==sizeof(int32_t); the fallback resampler does
not do any math on the sample values, so this hack is
possible: */
return (const float *)
pcm_resample_fallback_32(buffer, channels,
src_rate, (const int32_t *)src_buffer,
src_size,
dest_rate, dest_size_r);
}
const int16_t *
PcmResampler::Resample16(unsigned channels,
unsigned src_rate, const int16_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r)
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
return pcm_resample_lsr_16(this, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r,
error_r);
#else
(void)error_r;
#endif
return pcm_resample_fallback_16(buffer, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r);
}
const int32_t *
PcmResampler::Resample32(unsigned channels, unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r)
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
return pcm_resample_lsr_32(this, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r,
error_r);
#else
(void)error_r;
#endif
return pcm_resample_fallback_32(buffer, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r);
}
const int32_t *
PcmResampler::Resample24(unsigned channels, unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r)
{
#ifdef HAVE_LIBSAMPLERATE
if (pcm_resample_lsr_enabled())
return pcm_resample_lsr_24(this, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r,
error_r);
#else
(void)error_r;
#endif
/* reuse the 32 bit code - the resampler code doesn't care if
the upper 8 bits are actually used */
return pcm_resample_fallback_32(buffer, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r);
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_PCM_RESAMPLE_HXX
#define MPD_PCM_RESAMPLE_HXX
#include "check.h"
#include "PcmBuffer.hxx"
#include <stdint.h>
#include <stddef.h>
#ifdef HAVE_LIBSAMPLERATE
#include <samplerate.h>
#endif
class Error;
/**
* This object is statically allocated (within another struct), and
* holds buffer allocations and the state for the resampler.
*/
struct PcmResampler {
#ifdef HAVE_LIBSAMPLERATE
SRC_STATE *state;
SRC_DATA data;
PcmBuffer in, out;
struct {
unsigned src_rate;
unsigned dest_rate;
unsigned channels;
} prev;
int error;
#endif
PcmBuffer buffer;
PcmResampler();
~PcmResampler();
/**
* @see pcm_convert_reset()
*/
void Reset();
/**
* Resamples 32 bit float data.
*
* @param channels the number of channels
* @param src_rate the source sample rate
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_rate the requested destination sample rate
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer
*/
const float *ResampleFloat(unsigned channels, unsigned src_rate,
const float *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r);
/**
* Resamples 16 bit PCM data.
*
* @param channels the number of channels
* @param src_rate the source sample rate
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_rate the requested destination sample rate
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer
*/
const int16_t *Resample16(unsigned channels, unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r);
/**
* Resamples 32 bit PCM data.
*
* @param channels the number of channels
* @param src_rate the source sample rate
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_rate the requested destination sample rate
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer
*/
const int32_t *Resample32(unsigned channels, unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r);
/**
* Resamples 24 bit PCM data.
*
* @param channels the number of channels
* @param src_rate the source sample rate
* @param src the source PCM buffer
* @param src_size the size of #src in bytes
* @param dest_rate the requested destination sample rate
* @param dest_size_r returns the number of bytes of the destination buffer
* @return the destination buffer
*/
const int32_t *Resample24(unsigned channels, unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error_r);
};
bool
pcm_resample_global_init(Error &error);
#endif
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "PcmResampleInternal.hxx"
#include <assert.h>
/* resampling code blatantly ripped from ESD */
const int16_t *
pcm_resample_fallback_16(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate,
size_t *dest_size_r)
{
unsigned dest_pos = 0;
unsigned src_frames = src_size / channels / sizeof(*src_buffer);
unsigned dest_frames =
(src_frames * dest_rate + src_rate - 1) / src_rate;
unsigned dest_samples = dest_frames * channels;
size_t dest_size = dest_samples * sizeof(*src_buffer);
int16_t *dest_buffer = (int16_t *)buffer.Get(dest_size);
assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
switch (channels) {
case 1:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
dest_buffer[dest_pos++] = src_buffer[src_pos];
}
break;
case 2:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
src_pos &= ~1;
dest_buffer[dest_pos++] = src_buffer[src_pos];
dest_buffer[dest_pos++] = src_buffer[src_pos + 1];
}
break;
}
*dest_size_r = dest_size;
return dest_buffer;
}
const int32_t *
pcm_resample_fallback_32(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate,
size_t *dest_size_r)
{
unsigned dest_pos = 0;
unsigned src_frames = src_size / channels / sizeof(*src_buffer);
unsigned dest_frames =
(src_frames * dest_rate + src_rate - 1) / src_rate;
unsigned dest_samples = dest_frames * channels;
size_t dest_size = dest_samples * sizeof(*src_buffer);
int32_t *dest_buffer = (int32_t *)buffer.Get(dest_size);
assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
switch (channels) {
case 1:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
dest_buffer[dest_pos++] = src_buffer[src_pos];
}
break;
case 2:
while (dest_pos < dest_samples) {
unsigned src_pos = dest_pos * src_rate / dest_rate;
src_pos &= ~1;
dest_buffer[dest_pos++] = src_buffer[src_pos];
dest_buffer[dest_pos++] = src_buffer[src_pos + 1];
}
break;
}
*dest_size_r = dest_size;
return dest_buffer;
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
/** \file
*
* Internal declarations for the pcm_resample library. The "internal"
* resampler is called "fallback" in the MPD source, so the file name
* of this header is somewhat unrelated to it.
*/
#ifndef MPD_PCM_RESAMPLE_INTERNAL_HXX
#define MPD_PCM_RESAMPLE_INTERNAL_HXX
#include "check.h"
#include "PcmResample.hxx"
#ifdef HAVE_LIBSAMPLERATE
bool
pcm_resample_lsr_global_init(const char *converter, Error &error);
void
pcm_resample_lsr_init(PcmResampler *state);
void
pcm_resample_lsr_deinit(PcmResampler *state);
void
pcm_resample_lsr_reset(PcmResampler *state);
const float *
pcm_resample_lsr_float(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const float *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error);
const int16_t *
pcm_resample_lsr_16(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error);
const int32_t *
pcm_resample_lsr_32(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer,
size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error);
const int32_t *
pcm_resample_lsr_24(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer,
size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error);
#endif
const int16_t *
pcm_resample_fallback_16(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate,
size_t *dest_size_r);
const int32_t *
pcm_resample_fallback_32(PcmBuffer &buffer,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer,
size_t src_size,
unsigned dest_rate,
size_t *dest_size_r);
#endif
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "PcmResampleInternal.hxx"
#include "PcmUtils.hxx"
#include "util/ASCII.hxx"
#include "util/Error.hxx"
#include "util/Domain.hxx"
#include "Log.hxx"
#include <assert.h>
#include <stdlib.h>
#include <string.h>
static int lsr_converter = SRC_SINC_FASTEST;
static constexpr Domain libsamplerate_domain("libsamplerate");
static bool
lsr_parse_converter(const char *s)
{
assert(s != nullptr);
if (*s == 0)
return true;
char *endptr;
long l = strtol(s, &endptr, 10);
if (*endptr == 0 && src_get_name(l) != nullptr) {
lsr_converter = l;
return true;
}
size_t length = strlen(s);
for (int i = 0;; ++i) {
const char *name = src_get_name(i);
if (name == nullptr)
break;
if (StringEqualsCaseASCII(s, name, length)) {
lsr_converter = i;
return true;
}
}
return false;
}
bool
pcm_resample_lsr_global_init(const char *converter, Error &error)
{
if (!lsr_parse_converter(converter)) {
error.Format(libsamplerate_domain,
"unknown samplerate converter '%s'", converter);
return false;
}
FormatDebug(libsamplerate_domain,
"libsamplerate converter '%s'",
src_get_name(lsr_converter));
return true;
}
void
pcm_resample_lsr_init(PcmResampler *state)
{
state->state = nullptr;
memset(&state->data, 0, sizeof(state->data));
memset(&state->prev, 0, sizeof(state->prev));
state->error = 0;
}
void
pcm_resample_lsr_deinit(PcmResampler *state)
{
if (state->state != nullptr)
state->state = src_delete(state->state);
}
void
pcm_resample_lsr_reset(PcmResampler *state)
{
if (state->state != nullptr)
src_reset(state->state);
}
static bool
pcm_resample_set(PcmResampler *state,
unsigned channels, unsigned src_rate, unsigned dest_rate,
Error &error_r)
{
/* (re)set the state/ratio if the in or out format changed */
if (channels == state->prev.channels &&
src_rate == state->prev.src_rate &&
dest_rate == state->prev.dest_rate)
return true;
state->error = 0;
state->prev.channels = channels;
state->prev.src_rate = src_rate;
state->prev.dest_rate = dest_rate;
if (state->state)
state->state = src_delete(state->state);
int error;
state->state = src_new(lsr_converter, channels, &error);
if (!state->state) {
error_r.Format(libsamplerate_domain, error,
"libsamplerate initialization has failed: %s",
src_strerror(error));
return false;
}
SRC_DATA *data = &state->data;
data->src_ratio = (double)dest_rate / (double)src_rate;
FormatDebug(libsamplerate_domain,
"setting samplerate conversion ratio to %.2lf",
data->src_ratio);
src_set_ratio(state->state, data->src_ratio);
return true;
}
static bool
lsr_process(PcmResampler *state, Error &error)
{
if (state->error == 0)
state->error = src_process(state->state, &state->data);
if (state->error) {
error.Format(libsamplerate_domain, state->error,
"libsamplerate has failed: %s",
src_strerror(state->error));
return false;
}
return true;
}
const float *
pcm_resample_lsr_float(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const float *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error)
{
SRC_DATA *data = &state->data;
assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
if (!pcm_resample_set(state, channels, src_rate, dest_rate, error))
return nullptr;
data->input_frames = src_size / sizeof(*src_buffer) / channels;
data->data_in = const_cast<float *>(src_buffer);
data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
size_t data_out_size = data->output_frames * sizeof(float) * channels;
data->data_out = (float *)state->out.Get(data_out_size);
if (!lsr_process(state, error))
return nullptr;
*dest_size_r = data->output_frames_gen *
sizeof(*data->data_out) * channels;
return data->data_out;
}
const int16_t *
pcm_resample_lsr_16(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int16_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error)
{
SRC_DATA *data = &state->data;
assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
if (!pcm_resample_set(state, channels, src_rate, dest_rate,
error))
return nullptr;
data->input_frames = src_size / sizeof(*src_buffer) / channels;
size_t data_in_size = data->input_frames * sizeof(float) * channels;
data->data_in = (float *)state->in.Get(data_in_size);
data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
size_t data_out_size = data->output_frames * sizeof(float) * channels;
data->data_out = (float *)state->out.Get(data_out_size);
src_short_to_float_array(src_buffer, data->data_in,
data->input_frames * channels);
if (!lsr_process(state, error))
return nullptr;
int16_t *dest_buffer;
*dest_size_r = data->output_frames_gen *
sizeof(*dest_buffer) * channels;
dest_buffer = (int16_t *)state->buffer.Get(*dest_size_r);
src_float_to_short_array(data->data_out, dest_buffer,
data->output_frames_gen * channels);
return dest_buffer;
}
const int32_t *
pcm_resample_lsr_32(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error)
{
SRC_DATA *data = &state->data;
assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
if (!pcm_resample_set(state, channels, src_rate, dest_rate,
error))
return nullptr;
data->input_frames = src_size / sizeof(*src_buffer) / channels;
size_t data_in_size = data->input_frames * sizeof(float) * channels;
data->data_in = (float *)state->in.Get(data_in_size);
data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
size_t data_out_size = data->output_frames * sizeof(float) * channels;
data->data_out = (float *)state->out.Get(data_out_size);
src_int_to_float_array(src_buffer, data->data_in,
data->input_frames * channels);
if (!lsr_process(state, error))
return nullptr;
int32_t *dest_buffer;
*dest_size_r = data->output_frames_gen *
sizeof(*dest_buffer) * channels;
dest_buffer = (int32_t *)state->buffer.Get(*dest_size_r);
src_float_to_int_array(data->data_out, dest_buffer,
data->output_frames_gen * channels);
return dest_buffer;
}
const int32_t *
pcm_resample_lsr_24(PcmResampler *state,
unsigned channels,
unsigned src_rate,
const int32_t *src_buffer, size_t src_size,
unsigned dest_rate, size_t *dest_size_r,
Error &error)
{
const auto result = pcm_resample_lsr_32(state, channels,
src_rate, src_buffer, src_size,
dest_rate, dest_size_r,
error);
if (result != nullptr)
/* src_float_to_int_array() clamps for 32 bit
integers; now make sure everything's fine for 24
bit */
/* TODO: eliminate the 32 bit clamp to reduce overhead */
PcmClampN<int32_t, int32_t, 24>(const_cast<int32_t *>(result),
result,
*dest_size_r / sizeof(*result));
return result;
}
/*
* Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_PCM_RESAMPLER_HXX
#define MPD_PCM_RESAMPLER_HXX
#include "util/ConstBuffer.hxx"
#include "Compiler.h"
struct AudioFormat;
class Error;
/**
* This is an interface for plugins that convert PCM data to a
* specific sample rate.
*/
class PcmResampler {
public:
virtual ~PcmResampler() {}
/**
* Opens the resampler, preparing it for Resample().
*
* @param af the audio format of incoming data; the plugin may
* modify the object to enforce another input format (however,
* it may not request a different input sample rate)
* @param new_sample_rate the requested output sample rate
* @param error location to store the error
* @return the format of outgoing data or
* AudioFormat::Undefined() on error
*/
virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
Error &error) = 0;
/**
* Closes the resampler. After that, you may call Open()
* again.
*/
virtual void Close() = 0;
/**
* Resamples a block of PCM data.
*
* @param src the input buffer
* @param src_size the size of #src_buffer in bytes
* @param dest_size_r the size of the returned buffer
* @param error location to store the error occurring, or nullptr
* to ignore errors.
* @return the destination buffer on success (will be
* invalidated by filter_close() or filter_filter()), nullptr on
* error
*/
gcc_pure
virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
Error &error) = 0;
};
#endif
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment