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/*
* Copyright (C) 2003-2009 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_AUDIO_FORMAT_H
#define MPD_AUDIO_FORMAT_H
#include <stdint.h>
#include <stdbool.h>
/**
* This structure describes the format of a raw PCM stream.
*/
struct audio_format {
/**
* The sample rate in Hz. A better name for this attribute is
* "frame rate", because technically, you have two samples per
* frame in stereo sound.
*/
uint32_t sample_rate;
/**
* The number of significant bits per sample. Samples are
* currently always signed. Supported values are 8, 16, 24,
* 32. 24 bit samples are packed in 32 bit integers.
*/
uint8_t bits;
/**
* The number of channels. Only mono (1) and stereo (2) are
* fully supported currently.
*/
uint8_t channels;
/**
* If zero, then samples are stored in host byte order. If
* nonzero, then samples are stored in the reverse host byte
* order.
*/
uint8_t reverse_endian;
};
/**
* Clears the #audio_format object, i.e. sets all attributes to an
* undefined (invalid) value.
*/
static inline void audio_format_clear(struct audio_format *af)
{
af->sample_rate = 0;
af->bits = 0;
af->channels = 0;
af->reverse_endian = 0;
}
/**
* Initializes an #audio_format object, i.e. sets all
* attributes to valid values.
*/
static inline void audio_format_init(struct audio_format *af,
uint32_t sample_rate,
uint8_t bits, uint8_t channels)
{
af->sample_rate = sample_rate;
af->bits = bits;
af->channels = channels;
af->reverse_endian = 0;
}
/**
* Checks whether the specified #audio_format object has a defined
* value.
*/
static inline bool audio_format_defined(const struct audio_format *af)
{
return af->sample_rate != 0;
}
/**
* Checks whether the sample rate is valid.
*
* @param sample_rate the sample rate in Hz
*/
static inline bool
audio_valid_sample_rate(unsigned sample_rate)
{
return sample_rate > 0 && sample_rate < (1 << 30);
}
/**
* Checks whether the sample format is valid.
*
* @param bits the number of significant bits per sample
*/
static inline bool
audio_valid_sample_format(unsigned bits)
{
return bits == 16 || bits == 24 || bits == 32 || bits == 8;
}
/**
* Checks whether the number of channels is valid.
*/
static inline bool
audio_valid_channel_count(unsigned channels)
{
return channels >= 1 && channels <= 8;
}
/**
* Returns false if the format is not valid for playback with MPD.
* This function performs some basic validity checks.
*/
static inline bool audio_format_valid(const struct audio_format *af)
{
return audio_valid_sample_rate(af->sample_rate) &&
audio_valid_sample_format(af->bits) &&
audio_valid_channel_count(af->channels);
}
static inline bool audio_format_equals(const struct audio_format *a,
const struct audio_format *b)
{
return a->sample_rate == b->sample_rate &&
a->bits == b->bits &&
a->channels == b->channels &&
a->reverse_endian == b->reverse_endian;
}
/**
* Returns the size of each (mono) sample in bytes.
*/
static inline unsigned audio_format_sample_size(const struct audio_format *af)
{
if (af->bits <= 8)
return 1;
else if (af->bits <= 16)
return 2;
else
return 4;
}
/**
* Returns the size of each full frame in bytes.
*/
static inline unsigned
audio_format_frame_size(const struct audio_format *af)
{
return audio_format_sample_size(af) * af->channels;
}
/**
* Returns the floating point factor which converts a time span to a
* storage size in bytes.
*/
static inline double audio_format_time_to_size(const struct audio_format *af)
{
return af->sample_rate * audio_format_frame_size(af);
}
/**
* Returns the floating point factor which converts a storage size in
* bytes to a time span.
*/
static inline double audioFormatSizeToTime(const struct audio_format *af)
{
return 1.0 / audio_format_time_to_size(af);
}
#endif
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