/*
* Copyright (C) 2003-2011 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 <glib.h>
#include <stdint.h>
#include <stdbool.h>
#include <assert.h>
enum sample_format {
SAMPLE_FORMAT_UNDEFINED = 0,
SAMPLE_FORMAT_S8,
SAMPLE_FORMAT_S16,
/**
* Signed 24 bit integer samples, without padding.
*/
SAMPLE_FORMAT_S24,
/**
* Signed 24 bit integer samples, packed in 32 bit integers
* (the most significant byte is filled with the sign bit).
*/
SAMPLE_FORMAT_S24_P32,
SAMPLE_FORMAT_S32,
/**
* 32 bit floating point samples in the host's format. The
* range is -1.0f to +1.0f.
*/
SAMPLE_FORMAT_FLOAT,
/**
* Direct Stream Digital. 1-bit samples; each frame has one
* byte (8 samples) per channel.
*/
SAMPLE_FORMAT_DSD,
/**
* Same as #SAMPLE_FORMAT_DSD, but the least significant bit
* comes first.
*/
SAMPLE_FORMAT_DSD_LSBFIRST,
/**
* DSD packed in 24 bit samples (no padding), according to the
* dCS suggested standard:
* http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
*/
SAMPLE_FORMAT_DSD_OVER_USB,
};
static const unsigned MAX_CHANNELS = 8;
/**
* 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 format samples are stored in. See the #sample_format
* enum for valid values.
*/
uint8_t format;
/**
* 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.
*/
bool reverse_endian;
};
/**
* Buffer for audio_format_string().
*/
struct audio_format_string {
char buffer[24];
};
/**
* 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->format = SAMPLE_FORMAT_UNDEFINED;
af->channels = 0;
af->reverse_endian = false;
}
/**
* 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,
enum sample_format format, uint8_t channels)
{
af->sample_rate = sample_rate;
af->format = (uint8_t)format;
af->channels = channels;
af->reverse_endian = false;
}
/**
* 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 specified #audio_format object is full, i.e. all
* attributes are defined. This is more complete than
* audio_format_defined(), but slower.
*/
static inline bool
audio_format_fully_defined(const struct audio_format *af)
{
return af->sample_rate != 0 && af->format != SAMPLE_FORMAT_UNDEFINED &&
af->channels != 0;
}
/**
* Checks whether the specified #audio_format object has at least one
* defined value.
*/
static inline bool
audio_format_mask_defined(const struct audio_format *af)
{
return af->sample_rate != 0 || af->format != SAMPLE_FORMAT_UNDEFINED ||
af->channels != 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(enum sample_format format)
{
switch (format) {
case SAMPLE_FORMAT_S8:
case SAMPLE_FORMAT_S16:
case SAMPLE_FORMAT_S24:
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
case SAMPLE_FORMAT_FLOAT:
case SAMPLE_FORMAT_DSD:
case SAMPLE_FORMAT_DSD_LSBFIRST:
case SAMPLE_FORMAT_DSD_OVER_USB:
return true;
case SAMPLE_FORMAT_UNDEFINED:
break;
}
return false;
}
/**
* Checks whether the number of channels is valid.
*/
static inline bool
audio_valid_channel_count(unsigned channels)
{
return channels >= 1 && channels <= MAX_CHANNELS;
}
/**
* Returns false if the format is not valid for playback with MPD.
* This function performs some basic validity checks.
*/
G_GNUC_PURE
static inline bool audio_format_valid(const struct audio_format *af)
{
return audio_valid_sample_rate(af->sample_rate) &&
audio_valid_sample_format((enum sample_format)af->format) &&
audio_valid_channel_count(af->channels);
}
/**
* Returns false if the format mask is not valid for playback with
* MPD. This function performs some basic validity checks.
*/
G_GNUC_PURE
static inline bool audio_format_mask_valid(const struct audio_format *af)
{
return (af->sample_rate == 0 ||
audio_valid_sample_rate(af->sample_rate)) &&
(af->format == SAMPLE_FORMAT_UNDEFINED ||
audio_valid_sample_format((enum sample_format)af->format)) &&
(af->channels == 0 || 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->format == b->format &&
a->channels == b->channels &&
a->reverse_endian == b->reverse_endian;
}
void
audio_format_mask_apply(struct audio_format *af,
const struct audio_format *mask);
G_GNUC_CONST
static inline unsigned
sample_format_size(enum sample_format format)
{
switch (format) {
case SAMPLE_FORMAT_S8:
return 1;
case SAMPLE_FORMAT_S16:
return 2;
case SAMPLE_FORMAT_S24:
case SAMPLE_FORMAT_DSD_OVER_USB:
return 3;
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
case SAMPLE_FORMAT_FLOAT:
return 4;
case SAMPLE_FORMAT_DSD:
case SAMPLE_FORMAT_DSD_LSBFIRST:
/* each frame has 8 samples per channel */
return 1;
case SAMPLE_FORMAT_UNDEFINED:
return 0;
}
assert(false);
return 0;
}
/**
* Returns the size of each (mono) sample in bytes.
*/
G_GNUC_PURE
static inline unsigned audio_format_sample_size(const struct audio_format *af)
{
return sample_format_size((enum sample_format)af->format);
}
/**
* Returns the size of each full frame in bytes.
*/
G_GNUC_PURE
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.
*/
G_GNUC_PURE
static inline double audio_format_time_to_size(const struct audio_format *af)
{
return af->sample_rate * audio_format_frame_size(af);
}
/**
* Renders a #sample_format enum into a string, e.g. for printing it
* in a log file.
*
* @param format a #sample_format enum value
* @return the string
*/
G_GNUC_PURE G_GNUC_MALLOC
const char *
sample_format_to_string(enum sample_format format);
/**
* Renders the #audio_format object into a string, e.g. for printing
* it in a log file.
*
* @param af the #audio_format object
* @param s a buffer to print into
* @return the string, or NULL if the #audio_format object is invalid
*/
G_GNUC_PURE G_GNUC_MALLOC
const char *
audio_format_to_string(const struct audio_format *af,
struct audio_format_string *s);
#endif
