summaryrefslogtreecommitdiff
path: root/src/pcm/Export.hxx
blob: 8dfee34b6ca2eaa2487b8bf6d7a0ac903a77551a (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
/*
 * Copyright 2003-2019 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 "Buffer.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 #PcmConvert 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 DSD_U16 or DSD_U32.
	 *
	 * @see #dsd_u16, #dsd_u32
	 */
	PcmBuffer dsd_buffer;

	/**
	 * 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 empty (and 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