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/*
* Copyright 2003-2016 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 "check.h"
#include "PcmBuffer.hxx"
#include "AudioFormat.hxx"
template<typename T> struct ConstBuffer;
/**
* 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.
*/
struct PcmExport {
struct Params {
bool alsa_channel_order = false;
#ifdef ENABLE_DSD
bool dsd_u32 = false;
bool dop = false;
#endif
bool shift8 = false;
bool pack24 = false;
bool reverse_endian = false;
};
/**
* 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_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;
/**
* 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);
/**
* Calculate the size of one output frame.
*/
gcc_pure
size_t GetFrameSize(const AudioFormat &audio_format) const;
/**
* 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);
/**
* 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;
};
#endif
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