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-rw-r--r--src/audio_format.h205
1 files changed, 190 insertions, 15 deletions
diff --git a/src/audio_format.h b/src/audio_format.h
index 64087d070..dd32731c3 100644
--- a/src/audio_format.h
+++ b/src/audio_format.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
+ * Copyright (C) 2003-2010 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -23,25 +23,123 @@
#include <stdint.h>
#include <stdbool.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,
+};
+
+/**
+ * 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;
- uint8_t bits;
+
+ /**
+ * 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.
+ */
+ uint8_t 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->bits = 0;
+ af->format = SAMPLE_FORMAT_UNDEFINED;
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,
+ enum sample_format format, uint8_t channels)
+{
+ af->sample_rate = sample_rate;
+ af->format = (uint8_t)format;
+ 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 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
@@ -58,9 +156,21 @@ audio_valid_sample_rate(unsigned sample_rate)
* @param bits the number of significant bits per sample
*/
static inline bool
-audio_valid_sample_format(unsigned bits)
+audio_valid_sample_format(enum sample_format format)
{
- return bits == 16 || bits == 24 || bits == 32 || bits == 8;
+ switch (format) {
+ case SAMPLE_FORMAT_S8:
+ case SAMPLE_FORMAT_S16:
+ case SAMPLE_FORMAT_S24:
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
+ return true;
+
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+ }
+
+ return false;
}
/**
@@ -79,16 +189,44 @@ audio_valid_channel_count(unsigned channels)
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_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.
+ */
+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->bits == b->bits &&
- a->channels == b->channels;
+ a->format == b->format &&
+ a->channels == b->channels &&
+ a->reverse_endian == b->reverse_endian;
+}
+
+static inline void
+audio_format_mask_apply(struct audio_format *af,
+ const struct audio_format *mask)
+{
+ if (mask->sample_rate != 0)
+ af->sample_rate = mask->sample_rate;
+
+ if (mask->format != SAMPLE_FORMAT_UNDEFINED)
+ af->format = mask->format;
+
+ if (mask->channels != 0)
+ af->channels = mask->channels;
}
/**
@@ -96,28 +234,65 @@ static inline bool audio_format_equals(const struct audio_format *a,
*/
static inline unsigned audio_format_sample_size(const struct audio_format *af)
{
- if (af->bits <= 8)
+ switch (af->format) {
+ case SAMPLE_FORMAT_S8:
return 1;
- else if (af->bits <= 16)
+
+ case SAMPLE_FORMAT_S16:
return 2;
- else
+
+ case SAMPLE_FORMAT_S24:
+ return 3;
+
+ case SAMPLE_FORMAT_S24_P32:
+ case SAMPLE_FORMAT_S32:
return 4;
+
+ case SAMPLE_FORMAT_UNDEFINED:
+ break;
+ }
+
+ return 0;
}
+/**
+ * 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);
}
-static inline double audioFormatSizeToTime(const struct audio_format *af)
-{
- return 1.0 / audio_format_time_to_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
+ */
+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
+ */
+const char *
+audio_format_to_string(const struct audio_format *af,
+ struct audio_format_string *s);
#endif