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-rw-r--r--src/audio_format.h307
1 files changed, 0 insertions, 307 deletions
diff --git a/src/audio_format.h b/src/audio_format.h
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--- a/src/audio_format.h
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-/*
- * 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, 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,
-};
-
-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;
-};
-
-/**
- * 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;
-}
-
-/**
- * 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;
-}
-
-/**
- * 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_P32:
- case SAMPLE_FORMAT_S32:
- case SAMPLE_FORMAT_FLOAT:
- case SAMPLE_FORMAT_DSD:
- 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;
-}
-
-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_P32:
- case SAMPLE_FORMAT_S32:
- case SAMPLE_FORMAT_FLOAT:
- return 4;
-
- case SAMPLE_FORMAT_DSD:
- /* 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