diff options
Diffstat (limited to 'src/pcm')
41 files changed, 2164 insertions, 1681 deletions
diff --git a/src/pcm/ChannelsConverter.cxx b/src/pcm/ChannelsConverter.cxx new file mode 100644 index 000000000..8ffcbfe41 --- /dev/null +++ b/src/pcm/ChannelsConverter.cxx @@ -0,0 +1,97 @@ +/* + * 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 "ChannelsConverter.hxx" +#include "PcmChannels.hxx" +#include "Domain.hxx" +#include "util/ConstBuffer.hxx" +#include "util/Error.hxx" + +#include <assert.h> + +bool +PcmChannelsConverter::Open(SampleFormat _format, + unsigned _src_channels, unsigned _dest_channels, + gcc_unused Error &error) +{ + assert(_format != SampleFormat::UNDEFINED); + + switch (_format) { + case SampleFormat::S16: + case SampleFormat::S24_P32: + case SampleFormat::S32: + case SampleFormat::FLOAT: + break; + + default: + error.Format(pcm_domain, + "PCM channel conversion for %s is not implemented", + sample_format_to_string(format)); + return false; + } + + format = _format; + src_channels = _src_channels; + dest_channels = _dest_channels; + return true; +} + +void +PcmChannelsConverter::Close() +{ +#ifndef NDEBUG + format = SampleFormat::UNDEFINED; +#endif +} + +ConstBuffer<void> +PcmChannelsConverter::Convert(ConstBuffer<void> src, gcc_unused Error &error) +{ + switch (format) { + case SampleFormat::UNDEFINED: + case SampleFormat::S8: + case SampleFormat::DSD: + assert(false); + gcc_unreachable(); + + case SampleFormat::S16: + return pcm_convert_channels_16(buffer, dest_channels, + src_channels, + ConstBuffer<int16_t>::FromVoid(src)).ToVoid(); + + case SampleFormat::S24_P32: + return pcm_convert_channels_24(buffer, dest_channels, + src_channels, + ConstBuffer<int32_t>::FromVoid(src)).ToVoid(); + + case SampleFormat::S32: + return pcm_convert_channels_32(buffer, dest_channels, + src_channels, + ConstBuffer<int32_t>::FromVoid(src)).ToVoid(); + + case SampleFormat::FLOAT: + return pcm_convert_channels_float(buffer, dest_channels, + src_channels, + ConstBuffer<float>::FromVoid(src)).ToVoid(); + } + + assert(false); + gcc_unreachable(); +} diff --git a/src/pcm/ChannelsConverter.hxx b/src/pcm/ChannelsConverter.hxx new file mode 100644 index 000000000..4311b9671 --- /dev/null +++ b/src/pcm/ChannelsConverter.hxx @@ -0,0 +1,83 @@ +/* + * 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_CHANNELS_CONVERTER_HXX +#define MPD_PCM_CHANNELS_CONVERTER_HXX + +#include "check.h" +#include "AudioFormat.hxx" +#include "PcmBuffer.hxx" + +#ifndef NDEBUG +#include <assert.h> +#endif + +class Error; +template<typename T> struct ConstBuffer; + +/** + * A class that converts samples from one format to another. + */ +class PcmChannelsConverter { + SampleFormat format; + unsigned src_channels, dest_channels; + + PcmBuffer buffer; + +public: +#ifndef NDEBUG + PcmChannelsConverter() + :format(SampleFormat::UNDEFINED) {} + + ~PcmChannelsConverter() { + assert(format == SampleFormat::UNDEFINED); + } +#endif + + /** + * Opens the object, prepare for Convert(). + * + * @param format the sample format + * @param src_channels the number of source channels + * @param dest_channels the number of destination channels + * @param error location to store the error + * @return true on success + */ + bool Open(SampleFormat format, + unsigned src_channels, unsigned dest_channels, + Error &error); + + /** + * Closes the object. After that, you may call Open() again. + */ + void Close(); + + /** + * Convert a block of PCM data. + * + * @param src the input buffer + * @param error location to store the error + * @return the destination buffer on success, + * ConstBuffer::Null() on error + */ + gcc_pure + ConstBuffer<void> Convert(ConstBuffer<void> src, Error &error); +}; + +#endif diff --git a/src/pcm/ConfiguredResampler.cxx b/src/pcm/ConfiguredResampler.cxx new file mode 100644 index 000000000..845fa2332 --- /dev/null +++ b/src/pcm/ConfiguredResampler.cxx @@ -0,0 +1,101 @@ +/* + * 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" +#include "ConfigGlobal.hxx" +#include "ConfigOption.hxx" +#include "ConfigError.hxx" +#include "util/Error.hxx" + +#ifdef HAVE_LIBSAMPLERATE +#include "LibsamplerateResampler.hxx" +#endif + +#ifdef HAVE_SOXR +#include "SoxrResampler.hxx" +#endif + +#include <string.h> + +enum class SelectedResampler { + FALLBACK, + +#ifdef HAVE_LIBSAMPLERATE + LIBSAMPLERATE, +#endif + +#ifdef HAVE_SOXR + SOXR, +#endif +}; + +static SelectedResampler selected_resampler = SelectedResampler::FALLBACK; + +bool +pcm_resampler_global_init(Error &error) +{ + const char *converter = + config_get_string(CONF_SAMPLERATE_CONVERTER, ""); + + if (strcmp(converter, "internal") == 0) + return true; + +#ifdef HAVE_SOXR + if (strcmp(converter, "soxr") == 0) { + selected_resampler = SelectedResampler::SOXR; + return true; + } +#endif + +#ifdef HAVE_LIBSAMPLERATE + selected_resampler = SelectedResampler::LIBSAMPLERATE; + return pcm_resample_lsr_global_init(converter, error); +#endif + + if (*converter == 0) + return true; + + error.Format(config_domain, + "The samplerate_converter '%s' is not available", + converter); + return false; +} + +PcmResampler * +pcm_resampler_create() +{ + switch (selected_resampler) { + case SelectedResampler::FALLBACK: + return new FallbackPcmResampler(); + +#ifdef HAVE_LIBSAMPLERATE + case SelectedResampler::LIBSAMPLERATE: + return new LibsampleratePcmResampler(); +#endif + +#ifdef HAVE_SOXR + case SelectedResampler::SOXR: + return new SoxrPcmResampler(); +#endif + } + + gcc_unreachable(); +} 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/Domain.cxx b/src/pcm/Domain.cxx new file mode 100644 index 000000000..9f07d80dd --- /dev/null +++ b/src/pcm/Domain.cxx @@ -0,0 +1,23 @@ +/* + * 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 "Domain.hxx" +#include "util/Domain.hxx" + +const Domain pcm_domain("pcm"); diff --git a/src/pcm/Domain.hxx b/src/pcm/Domain.hxx new file mode 100644 index 000000000..170d7406d --- /dev/null +++ b/src/pcm/Domain.hxx @@ -0,0 +1,27 @@ +/* + * 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 PCM_DOMAIN_HXX +#define PCM_DOMAIN_HXX + +class Domain; + +extern const Domain pcm_domain; + +#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/FormatConverter.cxx b/src/pcm/FormatConverter.cxx new file mode 100644 index 000000000..8886a8ab0 --- /dev/null +++ b/src/pcm/FormatConverter.cxx @@ -0,0 +1,104 @@ +/* + * 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 "FormatConverter.hxx" +#include "PcmFormat.hxx" +#include "Domain.hxx" +#include "util/ConstBuffer.hxx" +#include "util/Error.hxx" + +#include <assert.h> + +bool +PcmFormatConverter::Open(SampleFormat _src_format, SampleFormat _dest_format, + gcc_unused Error &error) +{ + assert(_src_format != SampleFormat::UNDEFINED); + assert(_dest_format != SampleFormat::UNDEFINED); + + src_format = _src_format; + dest_format = _dest_format; + return true; +} + +void +PcmFormatConverter::Close() +{ +#ifndef NDEBUG + src_format = SampleFormat::UNDEFINED; + dest_format = SampleFormat::UNDEFINED; +#endif +} + +ConstBuffer<void> +PcmFormatConverter::Convert(ConstBuffer<void> src, Error &error) +{ + const void *result = nullptr; + size_t result_size = 0; + + switch (dest_format) { + case SampleFormat::UNDEFINED: + assert(false); + gcc_unreachable(); + + case SampleFormat::S8: + case SampleFormat::DSD: + result = nullptr; + break; + + case SampleFormat::S16: + result = pcm_convert_to_16(buffer, dither, + src_format, + src.data, src.size, + &result_size); + break; + + case SampleFormat::S24_P32: + result = pcm_convert_to_24(buffer, + src_format, + src.data, src.size, + &result_size); + break; + + case SampleFormat::S32: + result = pcm_convert_to_32(buffer, + src_format, + src.data, src.size, + &result_size); + break; + + case SampleFormat::FLOAT: + result = pcm_convert_to_float(buffer, + src_format, + src.data, src.size, + &result_size); + break; + } + + if (result == nullptr) { + error.Format(pcm_domain, + "PCM conversion from %s to %s is not implemented", + sample_format_to_string(src_format), + sample_format_to_string(dest_format)); + return nullptr; + } + + return { result, result_size }; +} diff --git a/src/pcm/FormatConverter.hxx b/src/pcm/FormatConverter.hxx new file mode 100644 index 000000000..f5b13a0b0 --- /dev/null +++ b/src/pcm/FormatConverter.hxx @@ -0,0 +1,84 @@ +/* + * 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_FORMAT_CONVERTER_HXX +#define MPD_PCM_FORMAT_CONVERTER_HXX + +#include "check.h" +#include "AudioFormat.hxx" +#include "PcmBuffer.hxx" +#include "PcmDither.hxx" + +#ifndef NDEBUG +#include <assert.h> +#endif + +class Error; +template<typename T> struct ConstBuffer; + +/** + * A class that converts samples from one format to another. + */ +class PcmFormatConverter { + SampleFormat src_format, dest_format; + + PcmBuffer buffer; + PcmDither dither; + +public: +#ifndef NDEBUG + PcmFormatConverter() + :src_format(SampleFormat::UNDEFINED), + dest_format(SampleFormat::UNDEFINED) {} + + ~PcmFormatConverter() { + assert(src_format == SampleFormat::UNDEFINED); + assert(dest_format == SampleFormat::UNDEFINED); + } +#endif + + /** + * Opens the object, prepare for Convert(). + * + * @param src_format the sample format of incoming data + * @param dest_format the sample format of outgoing data + * @param error location to store the error + * @return true on success + */ + bool Open(SampleFormat src_format, SampleFormat dest_format, + Error &error); + + /** + * Closes the object. After that, you may call Open() again. + */ + void Close(); + + /** + * Convert a block of PCM data. + * + * @param src the input buffer + * @param error location to store the error + * @return the destination buffer on success, + * ConstBuffer::Null() on error + */ + gcc_pure + ConstBuffer<void> Convert(ConstBuffer<void> src, Error &error); +}; + +#endif diff --git a/src/pcm/GlueResampler.cxx b/src/pcm/GlueResampler.cxx new file mode 100644 index 000000000..ef80e08a5 --- /dev/null +++ b/src/pcm/GlueResampler.cxx @@ -0,0 +1,85 @@ +/* + * 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 "GlueResampler.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, + Error &error) +{ + 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() +{ + if (requested_sample_format != src_sample_format) + format_converter.Close(); + + resampler->Close(); +} + +ConstBuffer<void> +GluePcmResampler::Resample(ConstBuffer<void> src, Error &error) +{ + assert(!src.IsNull()); + + if (requested_sample_format != src_sample_format) { + src = format_converter.Convert(src, error); + if (src.IsNull()) + return nullptr; + } + + return resampler->Resample(src, error); +} diff --git a/src/pcm/GlueResampler.hxx b/src/pcm/GlueResampler.hxx new file mode 100644 index 000000000..7bd923bab --- /dev/null +++ b/src/pcm/GlueResampler.hxx @@ -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. + */ + +#ifndef MPD_GLUE_RESAMPLER_HXX +#define MPD_GLUE_RESAMPLER_HXX + +#include "check.h" +#include "AudioFormat.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 *const resampler; + + SampleFormat src_sample_format, requested_sample_format; + SampleFormat output_sample_format; + + /** + * 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); +}; + +#endif 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/PcmBuffer.cxx b/src/pcm/PcmBuffer.cxx index 578c579be..5af9a978d 100644 --- a/src/pcm/PcmBuffer.cxx +++ b/src/pcm/PcmBuffer.cxx @@ -19,7 +19,6 @@ #include "config.h" #include "PcmBuffer.hxx" -#include "poison.h" void * PcmBuffer::Get(size_t new_size) diff --git a/src/pcm/PcmBuffer.hxx b/src/pcm/PcmBuffer.hxx index 717e24938..44a3ebf99 100644 --- a/src/pcm/PcmBuffer.hxx +++ b/src/pcm/PcmBuffer.hxx @@ -49,6 +49,12 @@ public: */ gcc_malloc void *Get(size_t size); + + template<typename T> + gcc_malloc + T *GetT(size_t n) { + return (T *)Get(n * sizeof(T)); + } }; #endif diff --git a/src/pcm/PcmChannels.cxx b/src/pcm/PcmChannels.cxx index eb69985c1..2e433e611 100644 --- a/src/pcm/PcmChannels.cxx +++ b/src/pcm/PcmChannels.cxx @@ -20,7 +20,9 @@ #include "config.h" #include "PcmChannels.hxx" #include "PcmBuffer.hxx" -#include "PcmUtils.hxx" +#include "Traits.hxx" +#include "AudioFormat.hxx" +#include "util/ConstBuffer.hxx" #include <assert.h> @@ -37,254 +39,143 @@ MonoToStereo(D dest, S src, S end) } -static void -pcm_convert_channels_16_2_to_1(int16_t *gcc_restrict dest, - const int16_t *gcc_restrict src, - const int16_t *gcc_restrict src_end) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::value_type +StereoToMono(typename Traits::value_type _a, + typename Traits::value_type _b) { - while (src < src_end) { - int32_t a = *src++, b = *src++; + typename Traits::sum_type a(_a); + typename Traits::sum_type b(_b); - *dest++ = (a + b) / 2; - } + return typename Traits::value_type((a + b) / 2); } -static void -pcm_convert_channels_16_n_to_2(int16_t *gcc_restrict dest, - unsigned src_channels, - const int16_t *gcc_restrict src, - const int16_t *gcc_restrict src_end) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::pointer_type +StereoToMono(typename Traits::pointer_type dest, + typename Traits::const_pointer_type src, + typename Traits::const_pointer_type end) { - unsigned c; - - assert(src_channels > 0); - - while (src < src_end) { - int32_t sum = 0; - int16_t value; - - for (c = 0; c < src_channels; ++c) - sum += *src++; - value = sum / (int)src_channels; + while (src != end) { + const auto a = *src++; + const auto b = *src++; - /* XXX this is actually only mono ... */ - *dest++ = value; - *dest++ = value; + *dest++ = StereoToMono<F, Traits>(a, b); } -} - -const int16_t * -pcm_convert_channels_16(PcmBuffer &buffer, - unsigned dest_channels, - unsigned src_channels, const int16_t *src, - size_t src_size, size_t *dest_size_r) -{ - assert(src_size % (sizeof(*src) * src_channels) == 0); - - size_t dest_size = src_size / src_channels * dest_channels; - *dest_size_r = dest_size; - - int16_t *dest = (int16_t *)buffer.Get(dest_size); - const int16_t *src_end = pcm_end_pointer(src, src_size); - - if (src_channels == 1 && dest_channels == 2) - MonoToStereo(dest, src, src_end); - else if (src_channels == 2 && dest_channels == 1) - pcm_convert_channels_16_2_to_1(dest, src, src_end); - else if (dest_channels == 2) - pcm_convert_channels_16_n_to_2(dest, src_channels, src, - src_end); - else - return nullptr; return dest; } -static void -pcm_convert_channels_24_2_to_1(int32_t *gcc_restrict dest, - const int32_t *gcc_restrict src, - const int32_t *gcc_restrict src_end) -{ - while (src < src_end) { - int32_t a = *src++, b = *src++; - - *dest++ = (a + b) / 2; - } -} - -static void -pcm_convert_channels_24_n_to_2(int32_t *gcc_restrict dest, - unsigned src_channels, - const int32_t *gcc_restrict src, - const int32_t *gcc_restrict src_end) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::pointer_type +NToStereo(typename Traits::pointer_type dest, + unsigned src_channels, + typename Traits::const_pointer_type src, + typename Traits::const_pointer_type end) { - unsigned c; + assert((end - src) % src_channels == 0); - assert(src_channels > 0); - - while (src < src_end) { - int32_t sum = 0; - int32_t value; - - for (c = 0; c < src_channels; ++c) + while (src != end) { + typename Traits::sum_type sum = *src++; + for (unsigned c = 1; c < src_channels; ++c) sum += *src++; - value = sum / (int)src_channels; - /* XXX this is actually only mono ... */ + typename Traits::value_type value(sum / int(src_channels)); + + /* TODO: this is actually only mono ... */ *dest++ = value; *dest++ = value; } -} - -const int32_t * -pcm_convert_channels_24(PcmBuffer &buffer, - unsigned dest_channels, - unsigned src_channels, const int32_t *src, - size_t src_size, size_t *dest_size_r) -{ - assert(src_size % (sizeof(*src) * src_channels) == 0); - - size_t dest_size = src_size / src_channels * dest_channels; - *dest_size_r = dest_size; - - int32_t *dest = (int32_t *)buffer.Get(dest_size); - const int32_t *src_end = (const int32_t *) - pcm_end_pointer(src, src_size); - - if (src_channels == 1 && dest_channels == 2) - MonoToStereo(dest, src, src_end); - else if (src_channels == 2 && dest_channels == 1) - pcm_convert_channels_24_2_to_1(dest, src, src_end); - else if (dest_channels == 2) - pcm_convert_channels_24_n_to_2(dest, src_channels, src, - src_end); - else - return nullptr; return dest; } -static void -pcm_convert_channels_32_2_to_1(int32_t *gcc_restrict dest, - const int32_t *gcc_restrict src, - const int32_t *gcc_restrict src_end) -{ - while (src < src_end) { - int64_t a = *src++, b = *src++; - - *dest++ = (a + b) / 2; - } -} - -static void -pcm_convert_channels_32_n_to_2(int32_t *dest, - unsigned src_channels, const int32_t *src, - const int32_t *src_end) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::pointer_type +NToM(typename Traits::pointer_type dest, + unsigned dest_channels, + unsigned src_channels, + typename Traits::const_pointer_type src, + typename Traits::const_pointer_type end) { - unsigned c; + assert((end - src) % src_channels == 0); - assert(src_channels > 0); - - while (src < src_end) { - int64_t sum = 0; - int32_t value; - - for (c = 0; c < src_channels; ++c) + while (src != end) { + typename Traits::sum_type sum = *src++; + for (unsigned c = 1; c < src_channels; ++c) sum += *src++; - value = sum / (int64_t)src_channels; - /* XXX this is actually only mono ... */ - *dest++ = value; - *dest++ = value; + typename Traits::value_type value(sum / int(src_channels)); + + /* TODO: this is actually only mono ... */ + for (unsigned c = 0; c < dest_channels; ++c) + *dest++ = value; } + + return dest; } -const int32_t * -pcm_convert_channels_32(PcmBuffer &buffer, - unsigned dest_channels, - unsigned src_channels, const int32_t *src, - size_t src_size, size_t *dest_size_r) +template<SampleFormat F, class Traits=SampleTraits<F>> +static ConstBuffer<typename Traits::value_type> +ConvertChannels(PcmBuffer &buffer, + unsigned dest_channels, + unsigned src_channels, + ConstBuffer<typename Traits::value_type> src) { - assert(src_size % (sizeof(*src) * src_channels) == 0); - - size_t dest_size = src_size / src_channels * dest_channels; - *dest_size_r = dest_size; + assert(src.size % src_channels == 0); - int32_t *dest = (int32_t *)buffer.Get(dest_size); - const int32_t *src_end = (const int32_t *) - pcm_end_pointer(src, src_size); + const size_t dest_size = src.size / src_channels * dest_channels; + auto dest = buffer.GetT<typename Traits::value_type>(dest_size); if (src_channels == 1 && dest_channels == 2) - MonoToStereo(dest, src, src_end); + MonoToStereo(dest, src.begin(), src.end()); else if (src_channels == 2 && dest_channels == 1) - pcm_convert_channels_32_2_to_1(dest, src, src_end); + StereoToMono<F>(dest, src.begin(), src.end()); else if (dest_channels == 2) - pcm_convert_channels_32_n_to_2(dest, src_channels, src, - src_end); + NToStereo<F>(dest, src_channels, src.begin(), src.end()); else - return nullptr; + NToM<F>(dest, dest_channels, + src_channels, src.begin(), src.end()); - return dest; + return { dest, dest_size }; } -static void -pcm_convert_channels_float_2_to_1(float *gcc_restrict dest, - const float *gcc_restrict src, - const float *gcc_restrict src_end) +ConstBuffer<int16_t> +pcm_convert_channels_16(PcmBuffer &buffer, + unsigned dest_channels, + unsigned src_channels, + ConstBuffer<int16_t> src) { - while (src < src_end) { - double a = *src++, b = *src++; - - *dest++ = (a + b) / 2; - } + return ConvertChannels<SampleFormat::S16>(buffer, dest_channels, + src_channels, src); } -static void -pcm_convert_channels_float_n_to_2(float *dest, - unsigned src_channels, const float *src, - const float *src_end) +ConstBuffer<int32_t> +pcm_convert_channels_24(PcmBuffer &buffer, + unsigned dest_channels, + unsigned src_channels, + ConstBuffer<int32_t> src) { - unsigned c; - - assert(src_channels > 0); - - while (src < src_end) { - double sum = 0; - float value; - - for (c = 0; c < src_channels; ++c) - sum += *src++; - value = sum / (double)src_channels; + return ConvertChannels<SampleFormat::S24_P32>(buffer, dest_channels, + src_channels, src); +} - /* XXX this is actually only mono ... */ - *dest++ = value; - *dest++ = value; - } +ConstBuffer<int32_t> +pcm_convert_channels_32(PcmBuffer &buffer, + unsigned dest_channels, + unsigned src_channels, + ConstBuffer<int32_t> src) +{ + return ConvertChannels<SampleFormat::S32>(buffer, dest_channels, + src_channels, src); } -const float * +ConstBuffer<float> pcm_convert_channels_float(PcmBuffer &buffer, unsigned dest_channels, - unsigned src_channels, const float *src, - size_t src_size, size_t *dest_size_r) + unsigned src_channels, + ConstBuffer<float> src) { - assert(src_size % (sizeof(*src) * src_channels) == 0); - - size_t dest_size = src_size / src_channels * dest_channels; - *dest_size_r = dest_size; - - float *dest = (float *)buffer.Get(dest_size); - const float *src_end = (const float *)pcm_end_pointer(src, src_size); - - if (src_channels == 1 && dest_channels == 2) - MonoToStereo(dest, src, src_end); - else if (src_channels == 2 && dest_channels == 1) - pcm_convert_channels_float_2_to_1(dest, src, src_end); - else if (dest_channels == 2) - pcm_convert_channels_float_n_to_2(dest, src_channels, src, - src_end); - else - return nullptr; - - return dest; + return ConvertChannels<SampleFormat::FLOAT>(buffer, dest_channels, + src_channels, src); } diff --git a/src/pcm/PcmChannels.hxx b/src/pcm/PcmChannels.hxx index c67822825..ca4e06a8c 100644 --- a/src/pcm/PcmChannels.hxx +++ b/src/pcm/PcmChannels.hxx @@ -24,6 +24,7 @@ #include <stddef.h> class PcmBuffer; +template<typename T> struct ConstBuffer; /** * Changes the number of channels in 16 bit PCM data. @@ -32,15 +33,13 @@ class PcmBuffer; * @param dest_channels the number of channels requested * @param src_channels the number of channels in the source buffer * @param src the source PCM buffer - * @param src_size the number of bytes in #src - * @param dest_size_r returns the number of bytes of the destination buffer * @return the destination buffer */ -const int16_t * +ConstBuffer<int16_t> pcm_convert_channels_16(PcmBuffer &buffer, unsigned dest_channels, - unsigned src_channels, const int16_t *src, - size_t src_size, size_t *dest_size_r); + unsigned src_channels, + ConstBuffer<int16_t> src); /** * Changes the number of channels in 24 bit PCM data (aligned at 32 @@ -50,15 +49,13 @@ pcm_convert_channels_16(PcmBuffer &buffer, * @param dest_channels the number of channels requested * @param src_channels the number of channels in the source buffer * @param src the source PCM buffer - * @param src_size the number of bytes in #src - * @param dest_size_r returns the number of bytes of the destination buffer * @return the destination buffer */ -const int32_t * +ConstBuffer<int32_t> pcm_convert_channels_24(PcmBuffer &buffer, unsigned dest_channels, - unsigned src_channels, const int32_t *src, - size_t src_size, size_t *dest_size_r); + unsigned src_channels, + ConstBuffer<int32_t> src); /** * Changes the number of channels in 32 bit PCM data. @@ -67,15 +64,13 @@ pcm_convert_channels_24(PcmBuffer &buffer, * @param dest_channels the number of channels requested * @param src_channels the number of channels in the source buffer * @param src the source PCM buffer - * @param src_size the number of bytes in #src - * @param dest_size_r returns the number of bytes of the destination buffer * @return the destination buffer */ -const int32_t * +ConstBuffer<int32_t> pcm_convert_channels_32(PcmBuffer &buffer, unsigned dest_channels, - unsigned src_channels, const int32_t *src, - size_t src_size, size_t *dest_size_r); + unsigned src_channels, + ConstBuffer<int32_t> src); /** * Changes the number of channels in 32 bit float PCM data. @@ -84,14 +79,12 @@ pcm_convert_channels_32(PcmBuffer &buffer, * @param dest_channels the number of channels requested * @param src_channels the number of channels in the source buffer * @param src the source PCM buffer - * @param src_size the number of bytes in #src - * @param dest_size_r returns the number of bytes of the destination buffer * @return the destination buffer */ -const float * +ConstBuffer<float> pcm_convert_channels_float(PcmBuffer &buffer, unsigned dest_channels, - unsigned src_channels, const float *src, - size_t src_size, size_t *dest_size_r); + unsigned src_channels, + ConstBuffer<float> src); #endif diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx index 8eafe527c..6eecd3b01 100644 --- a/src/pcm/PcmConvert.cxx +++ b/src/pcm/PcmConvert.cxx @@ -19,289 +19,151 @@ #include "config.h" #include "PcmConvert.hxx" -#include "PcmChannels.hxx" -#include "PcmFormat.hxx" +#include "Domain.hxx" +#include "ConfiguredResampler.hxx" #include "AudioFormat.hxx" +#include "util/ConstBuffer.hxx" #include "util/Error.hxx" #include "util/Domain.hxx" +#include "util/ConstBuffer.hxx" #include <assert.h> #include <math.h> -const Domain pcm_convert_domain("pcm_convert"); +bool +pcm_convert_global_init(Error &error) +{ + return pcm_resampler_global_init(error); +} PcmConvert::PcmConvert() { +#ifndef NDEBUG + src_format.Clear(); + dest_format.Clear(); +#endif } PcmConvert::~PcmConvert() { + assert(!src_format.IsValid()); + assert(!dest_format.IsValid()); } -void -PcmConvert::Reset() +bool +PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format, + Error &error) { - dsd.Reset(); - resampler.Reset(); -} + assert(!src_format.IsValid()); + assert(!dest_format.IsValid()); + assert(_src_format.IsValid()); + assert(_dest_format.IsValid()); -inline const int16_t * -PcmConvert::Convert16(const AudioFormat src_format, - const void *src_buffer, size_t src_size, - const AudioFormat dest_format, size_t *dest_size_r, - Error &error) -{ - const int16_t *buf; - size_t len; + src_format = _src_format; + dest_format = _dest_format; + + AudioFormat format = src_format; + if (format.format == SampleFormat::DSD) + format.format = SampleFormat::FLOAT; - assert(dest_format.format == SampleFormat::S16); + enable_resampler = format.sample_rate != dest_format.sample_rate; + if (enable_resampler) { + if (!resampler.Open(format, dest_format.sample_rate, error)) + return false; - buf = pcm_convert_to_16(format_buffer, dither, - src_format.format, - src_buffer, src_size, - &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %s to 16 bit is not implemented", - sample_format_to_string(src_format.format)); - return nullptr; + format.format = resampler.GetOutputSampleFormat(); + format.sample_rate = dest_format.sample_rate; } - if (src_format.channels != dest_format.channels) { - buf = pcm_convert_channels_16(channels_buffer, - dest_format.channels, - src_format.channels, - buf, len, &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %u to %u channels " - "is not implemented", - src_format.channels, - dest_format.channels); - return nullptr; - } + 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; } - if (src_format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample16(dest_format.channels, - src_format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return nullptr; + format.format = dest_format.format; + + enable_channels = format.channels != dest_format.channels; + if (enable_channels && + !channels_converter.Open(format.format, format.channels, + dest_format.channels, error)) { + if (enable_format) + format_converter.Close(); + if (enable_resampler) + resampler.Close(); + return false; } - *dest_size_r = len; - return buf; + return true; } -inline const int32_t * -PcmConvert::Convert24(const AudioFormat src_format, - const void *src_buffer, size_t src_size, - const AudioFormat dest_format, size_t *dest_size_r, - Error &error) +void +PcmConvert::Close() { - const int32_t *buf; - size_t len; - - assert(dest_format.format == SampleFormat::S24_P32); - - buf = pcm_convert_to_24(format_buffer, - src_format.format, - src_buffer, src_size, &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %s to 24 bit is not implemented", - sample_format_to_string(src_format.format)); - return nullptr; - } + if (enable_channels) + channels_converter.Close(); + if (enable_format) + format_converter.Close(); + if (enable_resampler) + resampler.Close(); - if (src_format.channels != dest_format.channels) { - buf = pcm_convert_channels_24(channels_buffer, - dest_format.channels, - src_format.channels, - buf, len, &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %u to %u channels " - "is not implemented", - src_format.channels, - dest_format.channels); - return nullptr; - } - } - - if (src_format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample24(dest_format.channels, - src_format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return nullptr; - } + dsd.Reset(); - *dest_size_r = len; - return buf; +#ifndef NDEBUG + src_format.Clear(); + dest_format.Clear(); +#endif } -inline const int32_t * -PcmConvert::Convert32(const AudioFormat src_format, - const void *src_buffer, size_t src_size, - const AudioFormat dest_format, size_t *dest_size_r, - Error &error) +const void * +PcmConvert::Convert(const void *src, size_t src_size, + size_t *dest_size_r, + Error &error) { - const int32_t *buf; - size_t len; - - assert(dest_format.format == SampleFormat::S32); - - buf = pcm_convert_to_32(format_buffer, - src_format.format, - src_buffer, src_size, &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %s to 32 bit is not implemented", - sample_format_to_string(src_format.format)); - return nullptr; - } - - if (src_format.channels != dest_format.channels) { - buf = pcm_convert_channels_32(channels_buffer, - dest_format.channels, - src_format.channels, - buf, len, &len); - if (buf == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %u to %u channels " - "is not implemented", - src_format.channels, - dest_format.channels); + ConstBuffer<void> buffer(src, src_size); + AudioFormat format = src_format; + + if (format.format == SampleFormat::DSD) { + auto s = ConstBuffer<uint8_t>::FromVoid(buffer); + auto d = dsd.ToFloat(format.channels, + false, s); + if (d.IsNull()) { + error.Set(pcm_domain, + "DSD to PCM conversion failed"); return nullptr; } - } - if (src_format.sample_rate != dest_format.sample_rate) { - buf = resampler.Resample32(dest_format.channels, - src_format.sample_rate, buf, len, - dest_format.sample_rate, &len, - error); - if (buf == nullptr) - return buf; + buffer = d.ToVoid(); + format.format = SampleFormat::FLOAT; } - *dest_size_r = len; - return buf; -} - -inline const float * -PcmConvert::ConvertFloat(const AudioFormat src_format, - const void *src_buffer, size_t src_size, - const AudioFormat dest_format, size_t *dest_size_r, - Error &error) -{ - const float *buffer = (const float *)src_buffer; - size_t size = src_size; - - assert(dest_format.format == SampleFormat::FLOAT); - - /* convert to float now */ + if (enable_resampler) { + buffer = resampler.Resample(buffer, error); + if (buffer.IsNull()) + return nullptr; - buffer = pcm_convert_to_float(format_buffer, - src_format.format, - buffer, size, &size); - if (buffer == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %s to float is not implemented", - sample_format_to_string(src_format.format)); - return nullptr; + format.format = resampler.GetOutputSampleFormat(); + format.sample_rate = dest_format.sample_rate; } - /* convert channels */ - - if (src_format.channels != dest_format.channels) { - buffer = pcm_convert_channels_float(channels_buffer, - dest_format.channels, - src_format.channels, - buffer, size, &size); - if (buffer == nullptr) { - error.Format(pcm_convert_domain, - "Conversion from %u to %u channels " - "is not implemented", - src_format.channels, - dest_format.channels); + if (enable_format) { + buffer = format_converter.Convert(buffer, error); + if (buffer.IsNull()) return nullptr; - } - } - /* resample with float, because this is the best format for - libsamplerate */ - - if (src_format.sample_rate != dest_format.sample_rate) { - buffer = resampler.ResampleFloat(dest_format.channels, - src_format.sample_rate, - buffer, size, - dest_format.sample_rate, - &size, error); - if (buffer == nullptr) - return nullptr; + format.format = dest_format.format; } - *dest_size_r = size; - return buffer; -} - -const void * -PcmConvert::Convert(AudioFormat src_format, - const void *src, size_t src_size, - const AudioFormat dest_format, - size_t *dest_size_r, - Error &error) -{ - AudioFormat float_format; - if (src_format.format == SampleFormat::DSD) { - size_t f_size; - const float *f = dsd.ToFloat(src_format.channels, - false, (const uint8_t *)src, - src_size, &f_size); - if (f == nullptr) { - error.Set(pcm_convert_domain, - "DSD to PCM conversion failed"); + if (enable_channels) { + buffer = channels_converter.Convert(buffer, error); + if (buffer.IsNull()) return nullptr; - } - float_format = src_format; - float_format.format = SampleFormat::FLOAT; - - src_format = float_format; - src = f; - src_size = f_size; + format.channels = dest_format.channels; } - switch (dest_format.format) { - case SampleFormat::S16: - return Convert16(src_format, src, src_size, - dest_format, dest_size_r, - error); - - case SampleFormat::S24_P32: - return Convert24(src_format, src, src_size, - dest_format, dest_size_r, - error); - - case SampleFormat::S32: - return Convert32(src_format, src, src_size, - dest_format, dest_size_r, - error); - - case SampleFormat::FLOAT: - return ConvertFloat(src_format, src, src_size, - dest_format, dest_size_r, - error); - - default: - error.Format(pcm_convert_domain, - "PCM conversion to %s is not implemented", - sample_format_to_string(dest_format.format)); - return nullptr; - } + *dest_size_r = buffer.size; + return buffer.data; } diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx index 40f785179..586c303f2 100644 --- a/src/pcm/PcmConvert.hxx +++ b/src/pcm/PcmConvert.hxx @@ -20,15 +20,18 @@ #ifndef PCM_CONVERT_HXX #define PCM_CONVERT_HXX -#include "PcmDither.hxx" #include "PcmDsd.hxx" -#include "PcmResample.hxx" #include "PcmBuffer.hxx" +#include "FormatConverter.hxx" +#include "ChannelsConverter.hxx" +#include "GlueResampler.hxx" +#include "AudioFormat.hxx" #include <stddef.h> -struct AudioFormat; +template<typename T> struct ConstBuffer; class Error; +class Domain; /** * This object is statically allocated (within another struct), and @@ -38,27 +41,29 @@ class Error; class PcmConvert { PcmDsd dsd; - PcmResampler resampler; + GluePcmResampler resampler; + PcmFormatConverter format_converter; + PcmChannelsConverter channels_converter; - PcmDither dither; + AudioFormat src_format, dest_format; - /** the buffer for converting the sample format */ - PcmBuffer format_buffer; - - /** the buffer for converting the channel count */ - PcmBuffer channels_buffer; + bool enable_resampler, enable_format, enable_channels; public: PcmConvert(); ~PcmConvert(); + /** + * Prepare the object. Call Close() when done. + */ + bool Open(AudioFormat _src_format, AudioFormat _dest_format, + Error &error); /** - * Reset the pcm_convert_state object. Use this at the - * boundary between two distinct songs and each time the - * format changes. + * Close the object after it was prepared with Open(). After + * that, it may be reused by calling Open() again. */ - void Reset(); + void Close(); /** * Converts PCM data between two audio formats. @@ -72,38 +77,12 @@ public: * ignore errors * @return the destination buffer, or NULL on error */ - const void *Convert(AudioFormat src_format, - const void *src, size_t src_size, - AudioFormat dest_format, + const void *Convert(const void *src, size_t src_size, size_t *dest_size_r, Error &error); - -private: - const int16_t *Convert16(AudioFormat src_format, - const void *src_buffer, size_t src_size, - AudioFormat dest_format, - size_t *dest_size_r, - Error &error); - - const int32_t *Convert24(AudioFormat src_format, - const void *src_buffer, size_t src_size, - AudioFormat dest_format, - size_t *dest_size_r, - Error &error); - - const int32_t *Convert32(AudioFormat src_format, - const void *src_buffer, size_t src_size, - AudioFormat dest_format, - size_t *dest_size_r, - Error &error); - - const float *ConvertFloat(AudioFormat src_format, - const void *src_buffer, size_t src_size, - AudioFormat dest_format, - size_t *dest_size_r, - Error &error); }; -extern const class Domain pcm_convert_domain; +bool +pcm_convert_global_init(Error &error); #endif diff --git a/src/pcm/PcmDither.cxx b/src/pcm/PcmDither.cxx index 98d0d443e..71ec5db1e 100644 --- a/src/pcm/PcmDither.cxx +++ b/src/pcm/PcmDither.cxx @@ -20,18 +20,14 @@ #include "config.h" #include "PcmDither.hxx" #include "PcmPrng.hxx" +#include "Traits.hxx" -inline int16_t -PcmDither::Dither24To16(int_fast32_t sample) +template<typename T, T MIN, T MAX, unsigned scale_bits> +inline T +PcmDither::Dither(T sample) { - constexpr unsigned from_bits = 24; - constexpr unsigned to_bits = 16; - constexpr unsigned scale_bits = from_bits - to_bits; - constexpr int_fast32_t round = 1 << (scale_bits - 1); - constexpr int_fast32_t mask = (1 << scale_bits) - 1; - constexpr int_fast32_t ONE = 1 << (from_bits - 1); - constexpr int_fast32_t MIN = -ONE; - constexpr int_fast32_t MAX = ONE - 1; + constexpr T round = 1 << (scale_bits - 1); + constexpr T mask = (1 << scale_bits) - 1; sample += error[0] - error[1] + error[2]; @@ -39,9 +35,9 @@ PcmDither::Dither24To16(int_fast32_t sample) error[1] = error[0] / 2; /* round */ - int_fast32_t output = sample + round; + T output = sample + round; - int_fast32_t rnd = pcm_prng(random); + const T rnd = pcm_prng(random); output += (rnd & mask) - (random & mask); random = rnd; @@ -63,27 +59,46 @@ PcmDither::Dither24To16(int_fast32_t sample) error[0] = sample - output; - return (int16_t)(output >> scale_bits); + return output; } -void -PcmDither::Dither24To16(int16_t *dest, const int32_t *src, - const int32_t *src_end) +template<typename ST, typename DT> +inline typename DT::value_type +PcmDither::DitherShift(typename ST::value_type sample) +{ + static_assert(ST::BITS > DT::BITS, + "Sample formats cannot be dithered"); + + constexpr unsigned scale_bits = ST::BITS - DT::BITS; + + return Dither<typename ST::sum_type, ST::MIN, ST::MAX, + scale_bits>(sample) >> scale_bits; +} + +template<typename ST, typename DT> +inline void +PcmDither::DitherShift(typename DT::pointer_type dest, + typename ST::const_pointer_type src, + typename ST::const_pointer_type src_end) { while (src < src_end) - *dest++ = Dither24To16(*src++); + *dest++ = DitherShift<ST, DT>(*src++); } -inline int16_t -PcmDither::Dither32To16(int_fast32_t sample) +void +PcmDither::Dither24To16(int16_t *dest, const int32_t *src, + const int32_t *src_end) { - return Dither24To16(sample >> 8); + typedef SampleTraits<SampleFormat::S24_P32> ST; + typedef SampleTraits<SampleFormat::S16> DT; + DitherShift<ST, DT>(dest, src, src_end); } void PcmDither::Dither32To16(int16_t *dest, const int32_t *src, const int32_t *src_end) { - while (src < src_end) - *dest++ = Dither32To16(*src++); + typedef SampleTraits<SampleFormat::S32> ST; + typedef SampleTraits<SampleFormat::S16> DT; + DitherShift<ST, DT>(dest, src, src_end); } diff --git a/src/pcm/PcmDither.hxx b/src/pcm/PcmDither.hxx index 106382307..2b75cdaeb 100644 --- a/src/pcm/PcmDither.hxx +++ b/src/pcm/PcmDither.hxx @@ -22,6 +22,8 @@ #include <stdint.h> +enum class SampleFormat : uint8_t; + class PcmDither { int32_t error[3]; int32_t random; @@ -37,8 +39,16 @@ public: const int32_t *src_end); private: - int16_t Dither24To16(int_fast32_t sample); - int16_t Dither32To16(int_fast32_t sample); + template<typename T, T MIN, T MAX, unsigned scale_bits> + T Dither(T sample); + + template<typename ST, typename DT> + typename DT::value_type DitherShift(typename ST::value_type sample); + + template<typename ST, typename DT> + void DitherShift(typename DT::pointer_type dest, + typename ST::const_pointer_type src, + typename ST::const_pointer_type src_end); }; #endif diff --git a/src/pcm/PcmDsd.cxx b/src/pcm/PcmDsd.cxx index 4db274635..5952cad7c 100644 --- a/src/pcm/PcmDsd.cxx +++ b/src/pcm/PcmDsd.cxx @@ -21,11 +21,11 @@ #include "PcmDsd.hxx" #include "dsd2pcm/dsd2pcm.h" #include "util/Macros.hxx" +#include "util/ConstBuffer.hxx" #include <algorithm> #include <assert.h> -#include <string.h> PcmDsd::PcmDsd() { @@ -47,22 +47,20 @@ PcmDsd::Reset() dsd2pcm_reset(dsd2pcm[i]); } -const float * +ConstBuffer<float> PcmDsd::ToFloat(unsigned channels, bool lsbfirst, - const uint8_t *src, size_t src_size, - size_t *dest_size_r) + ConstBuffer<uint8_t> src) { - assert(src != nullptr); - assert(src_size > 0); - assert(src_size % channels == 0); + assert(!src.IsNull()); + assert(!src.IsEmpty()); + assert(src.size % channels == 0); assert(channels <= ARRAY_SIZE(dsd2pcm)); - const unsigned num_samples = src_size; - const unsigned num_frames = src_size / channels; + const unsigned num_samples = src.size; + const unsigned num_frames = src.size / channels; float *dest; const size_t dest_size = num_samples * sizeof(*dest); - *dest_size_r = dest_size; dest = (float *)buffer.Get(dest_size); for (unsigned c = 0; c < channels; ++c) { @@ -73,9 +71,9 @@ PcmDsd::ToFloat(unsigned channels, bool lsbfirst, } dsd2pcm_translate(dsd2pcm[c], num_frames, - src + c, channels, + src.data + c, channels, lsbfirst, dest + c, channels); } - return dest; + return { dest, num_samples }; } diff --git a/src/pcm/PcmDsd.hxx b/src/pcm/PcmDsd.hxx index 26ee11b13..b9b6d51ee 100644 --- a/src/pcm/PcmDsd.hxx +++ b/src/pcm/PcmDsd.hxx @@ -25,22 +25,24 @@ #include <stdint.h> +template<typename T> struct ConstBuffer; + /** * Wrapper for the dsd2pcm library. */ -struct PcmDsd { +class PcmDsd { PcmBuffer buffer; struct dsd2pcm_ctx_s *dsd2pcm[32]; +public: PcmDsd(); ~PcmDsd(); void Reset(); - const float *ToFloat(unsigned channels, bool lsbfirst, - const uint8_t *src, size_t src_size, - size_t *dest_size_r); + ConstBuffer<float> ToFloat(unsigned channels, bool lsbfirst, + ConstBuffer<uint8_t> src); }; #endif diff --git a/src/pcm/PcmDsdUsb.cxx b/src/pcm/PcmDsdUsb.cxx index 2d0f33a15..833d84653 100644 --- a/src/pcm/PcmDsdUsb.cxx +++ b/src/pcm/PcmDsdUsb.cxx @@ -22,6 +22,8 @@ #include "PcmBuffer.hxx" #include "AudioFormat.hxx" +#include <assert.h> + constexpr static inline uint32_t pcm_two_dsd_to_usb_marker1(uint8_t a, uint8_t b) diff --git a/src/pcm/PcmFormat.cxx b/src/pcm/PcmFormat.cxx index 4565c71c6..a0e0bb2de 100644 --- a/src/pcm/PcmFormat.cxx +++ b/src/pcm/PcmFormat.cxx @@ -22,51 +22,7 @@ #include "PcmDither.hxx" #include "PcmBuffer.hxx" #include "PcmUtils.hxx" - -#include <type_traits> - -template<SampleFormat F> -struct SampleTraits {}; - -template<> -struct SampleTraits<SampleFormat::S8> { - typedef int8_t value_type; - typedef value_type *pointer_type; - typedef const value_type *const_pointer_type; - - static constexpr size_t SAMPLE_SIZE = sizeof(value_type); - static constexpr unsigned BITS = sizeof(value_type) * 8; -}; - -template<> -struct SampleTraits<SampleFormat::S16> { - typedef int16_t value_type; - typedef value_type *pointer_type; - typedef const value_type *const_pointer_type; - - static constexpr size_t SAMPLE_SIZE = sizeof(value_type); - static constexpr unsigned BITS = sizeof(value_type) * 8; -}; - -template<> -struct SampleTraits<SampleFormat::S32> { - typedef int32_t value_type; - typedef value_type *pointer_type; - typedef const value_type *const_pointer_type; - - static constexpr size_t SAMPLE_SIZE = sizeof(value_type); - static constexpr unsigned BITS = sizeof(value_type) * 8; -}; - -template<> -struct SampleTraits<SampleFormat::S24_P32> { - typedef int32_t value_type; - typedef value_type *pointer_type; - typedef const value_type *const_pointer_type; - - static constexpr size_t SAMPLE_SIZE = sizeof(value_type); - static constexpr unsigned BITS = 24; -}; +#include "Traits.hxx" static void pcm_convert_8_to_16(int16_t *out, const int8_t *in, const int8_t *in_end) @@ -100,8 +56,8 @@ ConvertFromFloat(typename Traits::pointer_type dest, const float factor = 1 << (bits - 1); while (src != end) { - int sample(*src++ * factor); - *dest++ = PcmClamp<typename Traits::value_type, int, bits>(sample); + typename Traits::long_type sample(*src++ * factor); + *dest++ = PcmClamp<F, Traits>(sample); } } @@ -442,7 +398,6 @@ 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/PcmMix.cxx b/src/pcm/PcmMix.cxx index 001794061..caf7fe516 100644 --- a/src/pcm/PcmMix.cxx +++ b/src/pcm/PcmMix.cxx @@ -19,42 +19,50 @@ #include "config.h" #include "PcmMix.hxx" -#include "PcmVolume.hxx" +#include "Volume.hxx" #include "PcmUtils.hxx" #include "AudioFormat.hxx" +#include "Traits.hxx" +#include "util/Clamp.hxx" +#include <assert.h> #include <math.h> -template<typename T, typename U, unsigned bits> -static T -PcmAddVolume(T _a, T _b, int volume1, int volume2) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::value_type +PcmAddVolume(typename Traits::value_type _a, typename Traits::value_type _b, + int volume1, int volume2) { - U a(_a), b(_b); + typename Traits::long_type a(_a), b(_b); - U c = ((a * volume1 + b * volume2) + - pcm_volume_dither() + PCM_VOLUME_1 / 2) - / PCM_VOLUME_1; + typename Traits::value_type c = ((a * volume1 + b * volume2) + + pcm_volume_dither() + PCM_VOLUME_1S / 2) + / PCM_VOLUME_1S; - return PcmClamp<T, U, bits>(c); + return PcmClamp<F, Traits>(c); } -template<typename T, typename U, unsigned bits> +template<SampleFormat F, class Traits=SampleTraits<F>> static void -PcmAddVolume(T *a, const T *b, unsigned n, int volume1, int volume2) +PcmAddVolume(typename Traits::pointer_type a, + typename Traits::const_pointer_type b, + size_t n, int volume1, int volume2) { for (size_t i = 0; i != n; ++i) - a[i] = PcmAddVolume<T, U, bits>(a[i], b[i], volume1, volume2); + a[i] = PcmAddVolume<F, Traits>(a[i], b[i], volume1, volume2); } -template<typename T, typename U, unsigned bits> +template<SampleFormat F, class Traits=SampleTraits<F>> static void PcmAddVolumeVoid(void *a, const void *b, size_t size, int volume1, int volume2) { - constexpr size_t sample_size = sizeof(T); + constexpr size_t sample_size = Traits::SAMPLE_SIZE; assert(size % sample_size == 0); - PcmAddVolume<T, U, bits>((T *)a, (const T *)b, size / sample_size, - volume1, volume2); + PcmAddVolume<F, Traits>(typename Traits::pointer_type(a), + typename Traits::const_pointer_type(b), + size / sample_size, + volume1, volume2); } static void @@ -83,23 +91,23 @@ pcm_add_vol(void *buffer1, const void *buffer2, size_t size, return false; case SampleFormat::S8: - PcmAddVolumeVoid<int8_t, int32_t, 8>(buffer1, buffer2, size, - vol1, vol2); + PcmAddVolumeVoid<SampleFormat::S8>(buffer1, buffer2, size, + vol1, vol2); return true; case SampleFormat::S16: - PcmAddVolumeVoid<int16_t, int32_t, 16>(buffer1, buffer2, size, - vol1, vol2); + PcmAddVolumeVoid<SampleFormat::S16>(buffer1, buffer2, size, + vol1, vol2); return true; case SampleFormat::S24_P32: - PcmAddVolumeVoid<int32_t, int64_t, 24>(buffer1, buffer2, size, - vol1, vol2); + PcmAddVolumeVoid<SampleFormat::S24_P32>(buffer1, buffer2, size, + vol1, vol2); return true; case SampleFormat::S32: - PcmAddVolumeVoid<int32_t, int64_t, 32>(buffer1, buffer2, size, - vol1, vol2); + PcmAddVolumeVoid<SampleFormat::S32>(buffer1, buffer2, size, + vol1, vol2); return true; case SampleFormat::FLOAT: @@ -114,30 +122,35 @@ pcm_add_vol(void *buffer1, const void *buffer2, size_t size, gcc_unreachable(); } -template<typename T, typename U, unsigned bits> -static T -PcmAdd(T _a, T _b) +template<SampleFormat F, class Traits=SampleTraits<F>> +static typename Traits::value_type +PcmAdd(typename Traits::value_type _a, typename Traits::value_type _b) { - U a(_a), b(_b); - return PcmClamp<T, U, bits>(a + b); + typename Traits::sum_type a(_a), b(_b); + + return PcmClamp<F, Traits>(a + b); } -template<typename T, typename U, unsigned bits> +template<SampleFormat F, class Traits=SampleTraits<F>> static void -PcmAdd(T *a, const T *b, unsigned n) +PcmAdd(typename Traits::pointer_type a, + typename Traits::const_pointer_type b, + size_t n) { for (size_t i = 0; i != n; ++i) - a[i] = PcmAdd<T, U, bits>(a[i], b[i]); + a[i] = PcmAdd<F, Traits>(a[i], b[i]); } -template<typename T, typename U, unsigned bits> +template<SampleFormat F, class Traits=SampleTraits<F>> static void PcmAddVoid(void *a, const void *b, size_t size) { - constexpr size_t sample_size = sizeof(T); + constexpr size_t sample_size = Traits::SAMPLE_SIZE; assert(size % sample_size == 0); - PcmAdd<T, U, bits>((T *)a, (const T *)b, size / sample_size); + PcmAdd<F, Traits>(typename Traits::pointer_type(a), + typename Traits::const_pointer_type(b), + size / sample_size); } static void @@ -162,19 +175,19 @@ pcm_add(void *buffer1, const void *buffer2, size_t size, return false; case SampleFormat::S8: - PcmAddVoid<int8_t, int32_t, 8>(buffer1, buffer2, size); + PcmAddVoid<SampleFormat::S8>(buffer1, buffer2, size); return true; case SampleFormat::S16: - PcmAddVoid<int16_t, int32_t, 16>(buffer1, buffer2, size); + PcmAddVoid<SampleFormat::S16>(buffer1, buffer2, size); return true; case SampleFormat::S24_P32: - PcmAddVoid<int32_t, int64_t, 24>(buffer1, buffer2, size); + PcmAddVoid<SampleFormat::S24_P32>(buffer1, buffer2, size); return true; case SampleFormat::S32: - PcmAddVoid<int32_t, int64_t, 32>(buffer1, buffer2, size); + PcmAddVoid<SampleFormat::S32>(buffer1, buffer2, size); return true; case SampleFormat::FLOAT: @@ -191,7 +204,6 @@ bool pcm_mix(void *buffer1, const void *buffer2, size_t size, SampleFormat format, float portion1) { - int vol1; float s; /* portion1 is between 0.0 and 1.0 for crossfading, MixRamp uses -1 @@ -202,8 +214,8 @@ pcm_mix(void *buffer1, const void *buffer2, size_t size, s = sin(M_PI_2 * portion1); s *= s; - vol1 = s * PCM_VOLUME_1 + 0.5; - vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1); + int vol1 = s * PCM_VOLUME_1S + 0.5; + vol1 = Clamp<int>(vol1, 0, PCM_VOLUME_1S); - return pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format); + return pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1S - vol1, format); } diff --git a/src/pcm/PcmPrng.hxx b/src/pcm/PcmPrng.hxx index 0c823250d..73b1456a8 100644 --- a/src/pcm/PcmPrng.hxx +++ b/src/pcm/PcmPrng.hxx @@ -24,7 +24,7 @@ * A very simple linear congruential PRNG. It's good enough for PCM * dithering. */ -static unsigned long +constexpr static inline unsigned long pcm_prng(unsigned long state) { return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL; diff --git a/src/pcm/PcmResample.cxx b/src/pcm/PcmResample.cxx deleted file mode 100644 index 01f269ea9..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(this, 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(this, 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(this, 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(this, 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 a62cd64f7..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(PcmResampler *state, - 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 *)state->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(PcmResampler *state, - 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 *)state->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 5090c13d1..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(PcmResampler *state, - 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(PcmResampler *state, - 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 9eac2d545..000000000 --- a/src/pcm/PcmResampleLibsamplerate.cxx +++ /dev/null @@ -1,310 +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; -} - -#ifdef HAVE_LIBSAMPLERATE_NOINT - -/* libsamplerate introduced these functions in v0.1.3 */ - -static void -src_int_to_float_array(const int *in, float *out, int len) -{ - while (len-- > 0) - *out++ = *in++ / (float)(1 << (24 - 1)); -} - -static void -src_float_to_int_array (const float *in, int *out, int len) -{ - while (len-- > 0) - *out++ = *in++ * (float)(1 << (24 - 1)); -} - -#endif - -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/PcmUtils.hxx b/src/pcm/PcmUtils.hxx index febe12d7b..0ca5dd4ac 100644 --- a/src/pcm/PcmUtils.hxx +++ b/src/pcm/PcmUtils.hxx @@ -26,6 +26,9 @@ #include <stdint.h> +enum class SampleFormat : uint8_t; +template<SampleFormat F> struct SampleTraits; + /** * Add a byte count to the specified pointer. This is a utility * function to convert a source pointer and a byte count to an "end" @@ -42,37 +45,24 @@ pcm_end_pointer(const T *p, size_t size) * Check if the value is within the range of the provided bit size, * and caps it if necessary. */ -template<typename T, typename U, unsigned bits> +template<SampleFormat F, class Traits=SampleTraits<F>> gcc_const -static inline T -PcmClamp(U x) +static inline typename Traits::value_type +PcmClamp(typename Traits::long_type x) { - constexpr U MIN_VALUE = -(U(1) << (bits - 1)); - constexpr U MAX_VALUE = (U(1) << (bits - 1)) - 1; + typedef typename Traits::value_type T; typedef std::numeric_limits<T> limits; - static_assert(MIN_VALUE >= limits::min(), "out of range"); - static_assert(MAX_VALUE <= limits::max(), "out of range"); + static_assert(Traits::MIN >= limits::min(), "out of range"); + static_assert(Traits::MAX <= limits::max(), "out of range"); - if (gcc_unlikely(x < MIN_VALUE)) - return T(MIN_VALUE); + if (gcc_unlikely(x < Traits::MIN)) + return T(Traits::MIN); - if (gcc_unlikely(x > MAX_VALUE)) - return T(MAX_VALUE); + if (gcc_unlikely(x > Traits::MAX)) + return T(Traits::MAX); return T(x); } -/** - * Check if the values in this buffer are within the range of the - * provided bit size, and clamps them whenever necessary. - */ -template<typename T, typename U, unsigned bits> -static inline void -PcmClampN(T *dest, const U *src, unsigned n) -{ - while (n-- > 0) - *dest++ = PcmClamp<T, U, bits>(*src++); -} - #endif diff --git a/src/pcm/PcmVolume.cxx b/src/pcm/PcmVolume.cxx deleted file mode 100644 index 564880633..000000000 --- a/src/pcm/PcmVolume.cxx +++ /dev/null @@ -1,188 +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 "PcmVolume.hxx" -#include "PcmUtils.hxx" -#include "AudioFormat.hxx" - -#include <stdint.h> -#include <string.h> - -static void -pcm_volume_change_8(int8_t *buffer, const int8_t *end, int volume) -{ - while (buffer < end) { - int32_t sample = *buffer; - - sample = (sample * volume + pcm_volume_dither() + - PCM_VOLUME_1 / 2) - / PCM_VOLUME_1; - - *buffer++ = PcmClamp<int8_t, int16_t, 8>(sample); - } -} - -static void -pcm_volume_change_16(int16_t *buffer, const int16_t *end, int volume) -{ - while (buffer < end) { - int32_t sample = *buffer; - - sample = (sample * volume + pcm_volume_dither() + - PCM_VOLUME_1 / 2) - / PCM_VOLUME_1; - - *buffer++ = PcmClamp<int16_t, int32_t, 16>(sample); - } -} - -#ifdef __i386__ -/** - * Optimized volume function for i386. Use the EDX:EAX 2*32 bit - * multiplication result instead of emulating 64 bit multiplication. - */ -static inline int32_t -pcm_volume_sample_24(int32_t sample, int32_t volume, gcc_unused int32_t dither) -{ - int32_t result; - - asm(/* edx:eax = sample * volume */ - "imul %2\n" - - /* "add %3, %1\n" dithering disabled for now, because we - have no overflow check - is dithering really important - here? */ - - /* eax = edx:eax / PCM_VOLUME_1 */ - "sal $22, %%edx\n" - "shr $10, %1\n" - "or %%edx, %1\n" - - : "=a"(result) - : "0"(sample), "r"(volume) /* , "r"(dither) */ - : "edx" - ); - - return result; -} -#endif - -static void -pcm_volume_change_24(int32_t *buffer, const int32_t *end, int volume) -{ - while (buffer < end) { -#ifdef __i386__ - /* assembly version for i386 */ - int32_t sample = *buffer; - - sample = pcm_volume_sample_24(sample, volume, - pcm_volume_dither()); -#else - /* portable version */ - int64_t sample = *buffer; - - sample = (sample * volume + pcm_volume_dither() + - PCM_VOLUME_1 / 2) - / PCM_VOLUME_1; -#endif - *buffer++ = PcmClamp<int32_t, int32_t, 24>(sample); - } -} - -static void -pcm_volume_change_32(int32_t *buffer, const int32_t *end, int volume) -{ - while (buffer < end) { -#ifdef __i386__ - /* assembly version for i386 */ - int32_t sample = *buffer; - - *buffer++ = pcm_volume_sample_24(sample, volume, 0); -#else - /* portable version */ - int64_t sample = *buffer; - - sample = (sample * volume + pcm_volume_dither() + - PCM_VOLUME_1 / 2) - / PCM_VOLUME_1; - *buffer++ = PcmClamp<int32_t, int64_t, 32>(sample); -#endif - } -} - -static void -pcm_volume_change_float(float *buffer, const float *end, float volume) -{ - while (buffer < end) { - float sample = *buffer; - sample *= volume; - *buffer++ = sample; - } -} - -bool -pcm_volume(void *buffer, size_t length, - SampleFormat format, - int volume) -{ - if (volume == PCM_VOLUME_1) - return true; - - if (volume <= 0) { - memset(buffer, 0, length); - return true; - } - - const void *end = pcm_end_pointer(buffer, length); - switch (format) { - case SampleFormat::UNDEFINED: - case SampleFormat::DSD: - /* not implemented */ - return false; - - case SampleFormat::S8: - pcm_volume_change_8((int8_t *)buffer, (const int8_t *)end, - volume); - return true; - - case SampleFormat::S16: - pcm_volume_change_16((int16_t *)buffer, (const int16_t *)end, - volume); - return true; - - case SampleFormat::S24_P32: - pcm_volume_change_24((int32_t *)buffer, (const int32_t *)end, - volume); - return true; - - case SampleFormat::S32: - pcm_volume_change_32((int32_t *)buffer, (const int32_t *)end, - volume); - return true; - - case SampleFormat::FLOAT: - pcm_volume_change_float((float *)buffer, (const float *)end, - pcm_volume_to_float(volume)); - return true; - } - - assert(false); - gcc_unreachable(); -} 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 diff --git a/src/pcm/SoxrResampler.cxx b/src/pcm/SoxrResampler.cxx new file mode 100644 index 000000000..e82ae1481 --- /dev/null +++ b/src/pcm/SoxrResampler.cxx @@ -0,0 +1,95 @@ +/* + * 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 "SoxrResampler.hxx" +#include "AudioFormat.hxx" +#include "util/ASCII.hxx" +#include "util/Error.hxx" +#include "util/Domain.hxx" +#include "Log.hxx" + +#include <soxr.h> + +#include <assert.h> + +static constexpr Domain soxr_domain("soxr"); + +AudioFormat +SoxrPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate, + Error &error) +{ + assert(af.IsValid()); + assert(audio_valid_sample_rate(new_sample_rate)); + + soxr_error_t e; + soxr = soxr_create(af.sample_rate, new_sample_rate, + af.channels, &e, + nullptr, nullptr, nullptr); + if (soxr == nullptr) { + error.Format(soxr_domain, + "soxr initialization has failed: %s", e); + return AudioFormat::Undefined(); + } + + FormatDebug(soxr_domain, "soxr engine '%s'", soxr_engine(soxr)); + + channels = af.channels; + + ratio = float(new_sample_rate) / float(af.sample_rate); + FormatDebug(soxr_domain, + "samplerate conversion ratio to %.2lf", + ratio); + + /* libsoxr works with floating point samples */ + af.format = SampleFormat::FLOAT; + + AudioFormat result = af; + result.sample_rate = new_sample_rate; + return result; +} + +void +SoxrPcmResampler::Close() +{ + soxr_delete(soxr); +} + +ConstBuffer<void> +SoxrPcmResampler::Resample(ConstBuffer<void> src, Error &error) +{ + const size_t frame_size = channels * sizeof(float); + assert(src.size % frame_size == 0); + + const size_t n_frames = src.size / frame_size; + + const size_t o_frames = size_t(n_frames * ratio + 0.5); + + float *output_buffer = (float *)buffer.Get(o_frames * frame_size); + + size_t i_done, o_done; + soxr_error_t e = soxr_process(soxr, src.data, n_frames, &i_done, + output_buffer, o_frames, &o_done); + if (e != nullptr) { + error.Format(soxr_domain, "soxr error: %s", e); + return nullptr; + } + + return { output_buffer, o_done * frame_size }; +} diff --git a/src/pcm/SoxrResampler.hxx b/src/pcm/SoxrResampler.hxx new file mode 100644 index 000000000..69c173741 --- /dev/null +++ b/src/pcm/SoxrResampler.hxx @@ -0,0 +1,47 @@ +/* + * 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_SOXR_RESAMPLER_HXX +#define MPD_PCM_SOXR_RESAMPLER_HXX + +#include "Resampler.hxx" +#include "PcmBuffer.hxx" + +struct AudioFormat; + +/** + * A resampler using soxr. + */ +class SoxrPcmResampler final : public PcmResampler { + struct soxr *soxr; + + unsigned channels; + float ratio; + + 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/Traits.hxx b/src/pcm/Traits.hxx new file mode 100644 index 000000000..4e68191f6 --- /dev/null +++ b/src/pcm/Traits.hxx @@ -0,0 +1,153 @@ +/* + * 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_TRAITS_HXX +#define MPD_PCM_TRAITS_HXX + +#include "check.h" +#include "AudioFormat.hxx" + +#include <stdint.h> +#include <stddef.h> + +/** + * This template describes the specified #SampleFormat. This is an + * empty prototype; the specializations contain the real definitions. + * See SampleTraits<uint8_t> for more documentation. + */ +template<SampleFormat F> +struct SampleTraits {}; + +template<> +struct SampleTraits<SampleFormat::S8> { + /** + * The type used for one sample value. + */ + typedef int8_t value_type; + + /** + * A writable pointer. + */ + typedef value_type *pointer_type; + + /** + * A read-only pointer. + */ + typedef const value_type *const_pointer_type; + + /** + * A "long" type that is large and accurate enough for adding + * two values without risking an (integer) overflow or + * (floating point) precision loss. + */ + typedef int sum_type; + + /** + * A "long" type that is large and accurate enough for + * arithmetic without risking an (integer) overflow or + * (floating point) precision loss. + */ + typedef int_least16_t long_type; + + /** + * The size of one sample in bytes. + */ + static constexpr size_t SAMPLE_SIZE = sizeof(value_type); + + /** + * The integer bit depth of one sample. This attribute may + * not exist if this is not an integer sample format. + */ + static constexpr unsigned BITS = sizeof(value_type) * 8; + + /** + * The minimum sample value. + */ + static constexpr value_type MIN = -(sum_type(1) << (BITS - 1)); + + /** + * The maximum sample value. + */ + static constexpr value_type MAX = (sum_type(1) << (BITS - 1)) - 1; +}; + +template<> +struct SampleTraits<SampleFormat::S16> { + typedef int16_t value_type; + typedef value_type *pointer_type; + typedef const value_type *const_pointer_type; + + typedef int_least32_t sum_type; + typedef int_least32_t long_type; + + static constexpr size_t SAMPLE_SIZE = sizeof(value_type); + static constexpr unsigned BITS = sizeof(value_type) * 8; + + static constexpr value_type MIN = -(sum_type(1) << (BITS - 1)); + static constexpr value_type MAX = (sum_type(1) << (BITS - 1)) - 1; +}; + +template<> +struct SampleTraits<SampleFormat::S32> { + typedef int32_t value_type; + typedef value_type *pointer_type; + typedef const value_type *const_pointer_type; + + typedef int_least64_t sum_type; + typedef int_least64_t long_type; + + static constexpr size_t SAMPLE_SIZE = sizeof(value_type); + static constexpr unsigned BITS = sizeof(value_type) * 8; + + static constexpr value_type MIN = -(sum_type(1) << (BITS - 1)); + static constexpr value_type MAX = (sum_type(1) << (BITS - 1)) - 1; +}; + +template<> +struct SampleTraits<SampleFormat::S24_P32> { + typedef int32_t value_type; + typedef value_type *pointer_type; + typedef const value_type *const_pointer_type; + + typedef int_least32_t sum_type; + typedef int_least64_t long_type; + + static constexpr size_t SAMPLE_SIZE = sizeof(value_type); + static constexpr unsigned BITS = 24; + + static constexpr value_type MIN = -(sum_type(1) << (BITS - 1)); + static constexpr value_type MAX = (sum_type(1) << (BITS - 1)) - 1; +}; + +template<> +struct SampleTraits<SampleFormat::FLOAT> { + typedef float value_type; + typedef value_type *pointer_type; + typedef const value_type *const_pointer_type; + + typedef float sum_type; + typedef float long_type; + + static constexpr size_t SAMPLE_SIZE = sizeof(value_type); + + static constexpr value_type MIN = -1; + static constexpr value_type MAX = 1; +}; + +#endif diff --git a/src/pcm/Volume.cxx b/src/pcm/Volume.cxx new file mode 100644 index 000000000..539ef1238 --- /dev/null +++ b/src/pcm/Volume.cxx @@ -0,0 +1,232 @@ +/* + * 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 "Volume.hxx" +#include "Domain.hxx" +#include "PcmUtils.hxx" +#include "Traits.hxx" +#include "util/ConstBuffer.hxx" +#include "util/Error.hxx" + +#include <stdint.h> +#include <string.h> + +template<SampleFormat F, class Traits=SampleTraits<F>> +static inline typename Traits::value_type +pcm_volume_sample(typename Traits::value_type _sample, + int volume) +{ + typename Traits::long_type sample(_sample); + + sample = (sample * volume + pcm_volume_dither() + + PCM_VOLUME_1S / 2) + >> PCM_VOLUME_BITS; + + return PcmClamp<F, Traits>(sample); +} + +#ifdef __i386__ + +/** + * Optimized volume function for i386. Use the EDX:EAX 2*32 bit + * multiplication result instead of emulating 64 bit multiplication. + */ +static inline int32_t +pcm_volume_sample_24(int32_t sample, int32_t volume, gcc_unused int32_t dither) +{ + int32_t result; + + asm(/* edx:eax = sample * volume */ + "imul %2\n" + + /* "add %3, %1\n" dithering disabled for now, because we + have no overflow check - is dithering really important + here? */ + + /* eax = edx:eax / PCM_VOLUME_1 */ + "sal $22, %%edx\n" + "shr $10, %1\n" + "or %%edx, %1\n" + + : "=a"(result) + : "0"(sample), "r"(volume) /* , "r"(dither) */ + : "edx" + ); + + return result; +} + +template<> +inline int32_t +pcm_volume_sample<SampleFormat::S24_P32, + SampleTraits<SampleFormat::S24_P32>>(int32_t sample, + int volume) +{ + return pcm_volume_sample_24(sample, volume, pcm_volume_dither()); +} + +template<> +inline int32_t +pcm_volume_sample<SampleFormat::S32, + SampleTraits<SampleFormat::S32>>(int32_t sample, int volume) +{ + return pcm_volume_sample_24(sample, volume, pcm_volume_dither()); +} + +#endif + +template<SampleFormat F, class Traits=SampleTraits<F>> +static void +pcm_volume_change(typename Traits::pointer_type dest, + typename Traits::const_pointer_type src, + typename Traits::const_pointer_type end, + int volume) +{ + while (src < end) { + const auto sample = *src++; + *dest++ = pcm_volume_sample<F, Traits>(sample, volume); + } +} + +static void +pcm_volume_change_8(int8_t *dest, const int8_t *src, const int8_t *end, + int volume) +{ + pcm_volume_change<SampleFormat::S8>(dest, src, end, volume); +} + +static void +pcm_volume_change_16(int16_t *dest, const int16_t *src, const int16_t *end, + int volume) +{ + pcm_volume_change<SampleFormat::S16>(dest, src, end, volume); +} + +static void +pcm_volume_change_24(int32_t *dest, const int32_t *src, const int32_t *end, + int volume) +{ + pcm_volume_change<SampleFormat::S24_P32>(dest, src, end, volume); +} + +static void +pcm_volume_change_32(int32_t *dest, const int32_t *src, const int32_t *end, + int volume) +{ + pcm_volume_change<SampleFormat::S32>(dest, src, end, volume); +} + +static void +pcm_volume_change_float(float *dest, const float *src, const float *end, + float volume) +{ + while (src < end) { + float sample = *src++; + sample *= volume; + *dest++ = sample; + } +} + +bool +PcmVolume::Open(SampleFormat _format, Error &error) +{ + assert(format == SampleFormat::UNDEFINED); + + switch (_format) { + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: + error.Format(pcm_domain, + "Software volume for %s is not implemented", + sample_format_to_string(_format)); + return false; + + case SampleFormat::S8: + case SampleFormat::S16: + case SampleFormat::S24_P32: + case SampleFormat::S32: + case SampleFormat::FLOAT: + break; + } + + format = _format; + return true; +} + +ConstBuffer<void> +PcmVolume::Apply(ConstBuffer<void> src) +{ + if (volume == PCM_VOLUME_1) + return src; + + void *data = buffer.Get(src.size); + + if (volume == 0) { + /* optimized special case: 0% volume = memset(0) */ + /* TODO: is this valid for all sample formats? What + about floating point? */ + memset(data, 0, src.size); + return { data, src.size }; + } + + const void *end = pcm_end_pointer(src.data, src.size); + switch (format) { + case SampleFormat::UNDEFINED: + case SampleFormat::DSD: + assert(false); + gcc_unreachable(); + + case SampleFormat::S8: + pcm_volume_change_8((int8_t *)data, + (const int8_t *)src.data, + (const int8_t *)end, + volume); + break; + + case SampleFormat::S16: + pcm_volume_change_16((int16_t *)data, + (const int16_t *)src.data, + (const int16_t *)end, + volume); + break; + + case SampleFormat::S24_P32: + pcm_volume_change_24((int32_t *)data, + (const int32_t *)src.data, + (const int32_t *)end, + volume); + break; + + case SampleFormat::S32: + pcm_volume_change_32((int32_t *)data, + (const int32_t *)src.data, + (const int32_t *)end, + volume); + break; + + case SampleFormat::FLOAT: + pcm_volume_change_float((float *)data, + (const float *)src.data, + (const float *)end, + pcm_volume_to_float(volume)); + break; + } + + return { data, src.size }; +} diff --git a/src/pcm/PcmVolume.hxx b/src/pcm/Volume.hxx index 8cd82acf7..c31aafb6e 100644 --- a/src/pcm/PcmVolume.hxx +++ b/src/pcm/Volume.hxx @@ -22,14 +22,28 @@ #include "PcmPrng.hxx" #include "AudioFormat.hxx" +#include "PcmBuffer.hxx" #include <stdint.h> #include <stddef.h> -enum { - /** this value means "100% volume" */ - PCM_VOLUME_1 = 1024, -}; +#ifndef NDEBUG +#include <assert.h> +#endif + +class Error; +template<typename T> struct ConstBuffer; + +/** + * Number of fractional bits for a fixed-point volume value. + */ +static constexpr unsigned PCM_VOLUME_BITS = 10; + +/** + * This value means "100% volume". + */ +static constexpr unsigned PCM_VOLUME_1 = 1024; +static constexpr int PCM_VOLUME_1S = PCM_VOLUME_1; struct AudioFormat; @@ -65,17 +79,66 @@ pcm_volume_dither(void) } /** - * Adjust the volume of the specified PCM buffer. - * - * @param buffer the PCM buffer - * @param length the length of the PCM buffer - * @param format the sample format of the PCM buffer - * @param volume the volume between 0 and #PCM_VOLUME_1 - * @return true on success, false if the audio format is not supported + * A class that converts samples from one format to another. */ -bool -pcm_volume(void *buffer, size_t length, - SampleFormat format, - int volume); +class PcmVolume { + SampleFormat format; + + unsigned volume; + + PcmBuffer buffer; + +public: + PcmVolume() + :volume(PCM_VOLUME_1) { +#ifndef NDEBUG + format = SampleFormat::UNDEFINED; +#endif + } + +#ifndef NDEBUG + ~PcmVolume() { + assert(format == SampleFormat::UNDEFINED); + } +#endif + + unsigned GetVolume() const { + return volume; + } + + /** + * @param _volume the volume level in the range + * [0..#PCM_VOLUME_1]; may be bigger than #PCM_VOLUME_1, but + * then it will most likely clip a lot + */ + void SetVolume(unsigned _volume) { + volume = _volume; + } + + /** + * Opens the object, prepare for Apply(). + * + * @param format the sample format + * @param error location to store the error + * @return true on success + */ + bool Open(SampleFormat format, Error &error); + + /** + * Closes the object. After that, you may call Open() again. + */ + void Close() { +#ifndef NDEBUG + assert(format != SampleFormat::UNDEFINED); + format = SampleFormat::UNDEFINED; +#endif + } + + /** + * Apply the volume level. + */ + gcc_pure + ConstBuffer<void> Apply(ConstBuffer<void> src); +}; #endif |