diff options
author | Max Kellermann <max@duempel.org> | 2013-11-11 22:31:46 +0100 |
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committer | Max Kellermann <max@duempel.org> | 2013-11-30 16:22:57 +0100 |
commit | 5ba90cd8ea87cf97d64409b7b4c59033c2450c77 (patch) | |
tree | eec1565c805de7bc142de82e5100cc3c1f86aa8d /src | |
parent | e9127523db55a267f67532fd61e913f2879324fc (diff) | |
download | mpd-5ba90cd8ea87cf97d64409b7b4c59033c2450c77.tar.gz mpd-5ba90cd8ea87cf97d64409b7b4c59033c2450c77.tar.xz mpd-5ba90cd8ea87cf97d64409b7b4c59033c2450c77.zip |
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.
Diffstat (limited to 'src')
-rw-r--r-- | src/pcm/ConfiguredResampler.cxx | 63 | ||||
-rw-r--r-- | src/pcm/ConfiguredResampler.hxx | 38 | ||||
-rw-r--r-- | src/pcm/FallbackResampler.cxx | 147 | ||||
-rw-r--r-- | src/pcm/FallbackResampler.hxx | 45 | ||||
-rw-r--r-- | src/pcm/GlueResampler.cxx | 99 | ||||
-rw-r--r-- | src/pcm/GlueResampler.hxx | 27 | ||||
-rw-r--r-- | src/pcm/LibsamplerateResampler.cxx | 163 | ||||
-rw-r--r-- | src/pcm/LibsamplerateResampler.hxx | 55 | ||||
-rw-r--r-- | src/pcm/PcmConvert.cxx | 62 | ||||
-rw-r--r-- | src/pcm/PcmConvert.hxx | 4 | ||||
-rw-r--r-- | src/pcm/PcmFormat.cxx | 1 | ||||
-rw-r--r-- | src/pcm/PcmResample.cxx | 173 | ||||
-rw-r--r-- | src/pcm/PcmResample.hxx | 133 | ||||
-rw-r--r-- | src/pcm/PcmResampleFallback.cxx | 106 | ||||
-rw-r--r-- | src/pcm/PcmResampleInternal.hxx | 100 | ||||
-rw-r--r-- | src/pcm/PcmResampleLibsamplerate.cxx | 290 | ||||
-rw-r--r-- | src/pcm/Resampler.hxx | 74 |
17 files changed, 694 insertions, 886 deletions
diff --git a/src/pcm/ConfiguredResampler.cxx b/src/pcm/ConfiguredResampler.cxx new file mode 100644 index 000000000..92f3d0903 --- /dev/null +++ b/src/pcm/ConfiguredResampler.cxx @@ -0,0 +1,63 @@ +/* + * 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(); +} diff --git a/src/pcm/ConfiguredResampler.hxx b/src/pcm/ConfiguredResampler.hxx new file mode 100644 index 000000000..6d12ee9c6 --- /dev/null +++ b/src/pcm/ConfiguredResampler.hxx @@ -0,0 +1,38 @@ +/* + * 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 diff --git a/src/pcm/FallbackResampler.cxx b/src/pcm/FallbackResampler.cxx new file mode 100644 index 000000000..a3b6b78ee --- /dev/null +++ b/src/pcm/FallbackResampler.cxx @@ -0,0 +1,147 @@ +/* + * 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(); +} diff --git a/src/pcm/FallbackResampler.hxx b/src/pcm/FallbackResampler.hxx new file mode 100644 index 000000000..1b8d0377d --- /dev/null +++ b/src/pcm/FallbackResampler.hxx @@ -0,0 +1,45 @@ +/* + * 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 diff --git a/src/pcm/GlueResampler.cxx b/src/pcm/GlueResampler.cxx index 8aee98f38..ef80e08a5 100644 --- a/src/pcm/GlueResampler.cxx +++ b/src/pcm/GlueResampler.cxx @@ -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); } diff --git a/src/pcm/GlueResampler.hxx b/src/pcm/GlueResampler.hxx index a7e0a84f2..7bd923bab 100644 --- a/src/pcm/GlueResampler.hxx +++ b/src/pcm/GlueResampler.hxx @@ -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); }; diff --git a/src/pcm/LibsamplerateResampler.cxx b/src/pcm/LibsamplerateResampler.cxx new file mode 100644 index 000000000..586391e67 --- /dev/null +++ b/src/pcm/LibsamplerateResampler.cxx @@ -0,0 +1,163 @@ +/* + * 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(); +} diff --git a/src/pcm/LibsamplerateResampler.hxx b/src/pcm/LibsamplerateResampler.hxx new file mode 100644 index 000000000..0c1f613c8 --- /dev/null +++ b/src/pcm/LibsamplerateResampler.hxx @@ -0,0 +1,55 @@ +/* + * 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 diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx index 0552962aa..5501d8ddf 100644 --- a/src/pcm/PcmConvert.cxx +++ b/src/pcm/PcmConvert.cxx @@ -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; diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx index 9835045d6..13dff4427 100644 --- a/src/pcm/PcmConvert.hxx +++ b/src/pcm/PcmConvert.hxx @@ -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(); diff --git a/src/pcm/PcmFormat.cxx b/src/pcm/PcmFormat.cxx index 4565c71c6..b7a496264 100644 --- a/src/pcm/PcmFormat.cxx +++ b/src/pcm/PcmFormat.cxx @@ -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>> diff --git a/src/pcm/PcmResample.cxx b/src/pcm/PcmResample.cxx deleted file mode 100644 index b30e01407..000000000 --- a/src/pcm/PcmResample.cxx +++ /dev/null @@ -1,173 +0,0 @@ -/* - * 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); -} diff --git a/src/pcm/PcmResample.hxx b/src/pcm/PcmResample.hxx deleted file mode 100644 index e839d6ecd..000000000 --- a/src/pcm/PcmResample.hxx +++ /dev/null @@ -1,133 +0,0 @@ -/* - * 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 diff --git a/src/pcm/PcmResampleFallback.cxx b/src/pcm/PcmResampleFallback.cxx deleted file mode 100644 index ca92e5a83..000000000 --- a/src/pcm/PcmResampleFallback.cxx +++ /dev/null @@ -1,106 +0,0 @@ -/* - * 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; -} diff --git a/src/pcm/PcmResampleInternal.hxx b/src/pcm/PcmResampleInternal.hxx deleted file mode 100644 index 94cef94ff..000000000 --- a/src/pcm/PcmResampleInternal.hxx +++ /dev/null @@ -1,100 +0,0 @@ -/* - * 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 diff --git a/src/pcm/PcmResampleLibsamplerate.cxx b/src/pcm/PcmResampleLibsamplerate.cxx deleted file mode 100644 index e61ff2edf..000000000 --- a/src/pcm/PcmResampleLibsamplerate.cxx +++ /dev/null @@ -1,290 +0,0 @@ -/* - * 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; -} diff --git a/src/pcm/Resampler.hxx b/src/pcm/Resampler.hxx new file mode 100644 index 000000000..a74ef4e77 --- /dev/null +++ b/src/pcm/Resampler.hxx @@ -0,0 +1,74 @@ +/* + * 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 |