audio_format: added API documentation

This commit is contained in:
Max Kellermann 2009-07-22 15:56:48 +02:00
parent 6a071efa27
commit 44c97a8f6d
1 changed files with 48 additions and 0 deletions

View File

@ -23,13 +23,42 @@
#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;
@ -38,6 +67,10 @@ static inline void audio_format_clear(struct audio_format *af)
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)
@ -48,6 +81,10 @@ static inline void audio_format_init(struct audio_format *af,
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;
@ -117,17 +154,28 @@ 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);
}
/**
* 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);