ce49d99c2f
Since we switched from autotools to Meson in commit
94592c1406
, we don't need to include
`config.h` early to properly enable large file support. Meson passes
the required macros on the compiler command line instead of defining
them in `config.h`.
This means we can include `config.h` at any time, whenever we want to
check its macros, and there are no ordering constraints.
190 lines
4.4 KiB
C++
190 lines
4.4 KiB
C++
/*
|
|
* Copyright 2003-2018 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_EXPORT_HXX
|
|
#define PCM_EXPORT_HXX
|
|
|
|
#include "SampleFormat.hxx"
|
|
#include "PcmBuffer.hxx"
|
|
#include "config.h"
|
|
|
|
template<typename T> struct ConstBuffer;
|
|
struct AudioFormat;
|
|
|
|
/**
|
|
* An object that handles export of PCM samples to some instance
|
|
* outside of MPD. It has a few more options to tweak the binary
|
|
* representation which are not supported by the pcm_convert library.
|
|
*/
|
|
class PcmExport {
|
|
/**
|
|
* This buffer is used to reorder channels.
|
|
*
|
|
* @see #alsa_channel_order
|
|
*/
|
|
PcmBuffer order_buffer;
|
|
|
|
#ifdef ENABLE_DSD
|
|
/**
|
|
* The buffer is used to convert DSD samples to the
|
|
* DoP format.
|
|
*
|
|
* @see #dop
|
|
*/
|
|
PcmBuffer dop_buffer;
|
|
#endif
|
|
|
|
/**
|
|
* The buffer is used to pack samples, removing padding.
|
|
*
|
|
* @see #pack24
|
|
*/
|
|
PcmBuffer pack_buffer;
|
|
|
|
/**
|
|
* The buffer is used to reverse the byte order.
|
|
*
|
|
* @see #reverse_endian
|
|
*/
|
|
PcmBuffer reverse_buffer;
|
|
|
|
/**
|
|
* The number of channels.
|
|
*/
|
|
uint8_t channels;
|
|
|
|
/**
|
|
* Convert the given buffer from FLAC channel order to ALSA
|
|
* channel order using ToAlsaChannelOrder()?
|
|
*
|
|
* If this value is SampleFormat::UNDEFINED, then no channel
|
|
* reordering is applied, otherwise this is the input sample
|
|
* format.
|
|
*/
|
|
SampleFormat alsa_channel_order;
|
|
|
|
#ifdef ENABLE_DSD
|
|
/**
|
|
* Convert DSD (U8) to DSD_U16?
|
|
*/
|
|
bool dsd_u16;
|
|
|
|
/**
|
|
* Convert DSD (U8) to DSD_U32?
|
|
*/
|
|
bool dsd_u32;
|
|
|
|
/**
|
|
* Convert DSD to DSD-over-PCM (DoP)? Input format must be
|
|
* SampleFormat::DSD and output format must be
|
|
* SampleFormat::S24_P32.
|
|
*/
|
|
bool dop;
|
|
#endif
|
|
|
|
/**
|
|
* Convert (padded) 24 bit samples to 32 bit by shifting 8
|
|
* bits to the left?
|
|
*/
|
|
bool shift8;
|
|
|
|
/**
|
|
* Pack 24 bit samples?
|
|
*/
|
|
bool pack24;
|
|
|
|
/**
|
|
* Export the samples in reverse byte order? A non-zero value
|
|
* means the option is enabled and represents the size of each
|
|
* sample (2 or bigger).
|
|
*/
|
|
uint8_t reverse_endian;
|
|
|
|
public:
|
|
struct Params {
|
|
bool alsa_channel_order = false;
|
|
#ifdef ENABLE_DSD
|
|
bool dsd_u16 = false;
|
|
bool dsd_u32 = false;
|
|
bool dop = false;
|
|
#endif
|
|
bool shift8 = false;
|
|
bool pack24 = false;
|
|
bool reverse_endian = false;
|
|
|
|
/**
|
|
* Calculate the output sample rate, given a specific input
|
|
* sample rate. Usually, both are the same; however, with
|
|
* DSD_U32, four input bytes (= 4 * 8 bits) are combined to
|
|
* one output word (32 bits), dividing the sample rate by 4.
|
|
*/
|
|
gcc_pure
|
|
unsigned CalcOutputSampleRate(unsigned input_sample_rate) const noexcept;
|
|
|
|
/**
|
|
* The inverse of CalcOutputSampleRate().
|
|
*/
|
|
gcc_pure
|
|
unsigned CalcInputSampleRate(unsigned output_sample_rate) const noexcept;
|
|
};
|
|
|
|
/**
|
|
* Open the object.
|
|
*
|
|
* There is no "close" method. This function may be called multiple
|
|
* times to reuse the object.
|
|
*
|
|
* This function cannot fail.
|
|
*
|
|
* @param channels the number of channels; ignored unless dop is set
|
|
*/
|
|
void Open(SampleFormat sample_format, unsigned channels,
|
|
Params params) noexcept;
|
|
|
|
/**
|
|
* Reset the filter's state, e.g. drop/flush buffers.
|
|
*/
|
|
void Reset() noexcept {
|
|
}
|
|
|
|
/**
|
|
* Calculate the size of one output frame.
|
|
*/
|
|
gcc_pure
|
|
size_t GetFrameSize(const AudioFormat &audio_format) const noexcept;
|
|
|
|
/**
|
|
* Export a PCM buffer.
|
|
*
|
|
* @param src the source PCM buffer
|
|
* @return the destination buffer (may be a pointer to the source buffer)
|
|
*/
|
|
ConstBuffer<void> Export(ConstBuffer<void> src) noexcept;
|
|
|
|
/**
|
|
* Converts the number of consumed bytes from the pcm_export()
|
|
* destination buffer to the according number of bytes from the
|
|
* pcm_export() source buffer.
|
|
*/
|
|
gcc_pure
|
|
size_t CalcSourceSize(size_t dest_size) const noexcept;
|
|
};
|
|
|
|
#endif
|