1 files changed, 123 insertions, 5 deletions

diff --git a/src/audio_format.h b/src/audio_format.h
index 64087d070..0c1e425a9 100644
--- a/src/audio_format.h
+++ b/src/audio_format.h
@@ -23,25 +23,102 @@
 #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;
 };
 
+/**
+ * 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->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 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->bits != 0 && 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->bits != 0 || af->channels != 0;
+}
+
+/**
  * Checks whether the sample rate is valid.
  *
  * @param sample_rate the sample rate in Hz
@@ -83,12 +160,39 @@ static inline bool audio_format_valid(const struct audio_format *af)
 		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->bits == 0 || audio_valid_sample_format(af->bits)) &&
+		(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->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->bits != 0)
+		af->bits = mask->bits;
+
+	if (mask->channels != 0)
+		af->channels = mask->channels;
 }
 
 /**
@@ -104,20 +208,34 @@ static inline unsigned audio_format_sample_size(const struct audio_format *af)
 		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);
 }
 
-static inline double audioFormatSizeToTime(const struct audio_format *af)
-{
-	return 1.0 / audio_format_time_to_size(af);
-}
+/**
+ * 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