diff options
Diffstat (limited to 'src/audio_format.h')
| -rw-r--r-- | src/audio_format.h | 93 |
1 files changed, 48 insertions, 45 deletions
diff --git a/src/audio_format.h b/src/audio_format.h index a4450ad71..bf77add3b 100644 --- a/src/audio_format.h +++ b/src/audio_format.h @@ -1,5 +1,5 @@ /* - * Copyright (C) 2003-2010 The Music Player Daemon Project + * 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 @@ -20,6 +20,7 @@ #ifndef MPD_AUDIO_FORMAT_H #define MPD_AUDIO_FORMAT_H +#include <glib.h> #include <stdint.h> #include <stdbool.h> #include <assert.h> @@ -31,19 +32,28 @@ enum sample_format { 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, }; +static const unsigned MAX_CHANNELS = 8; + /** * This structure describes the format of a raw PCM stream. */ @@ -66,13 +76,6 @@ struct audio_format { * 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; }; /** @@ -91,7 +94,6 @@ 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 = 0; } /** @@ -105,7 +107,6 @@ static inline void audio_format_init(struct audio_format *af, af->sample_rate = sample_rate; af->format = (uint8_t)format; af->channels = channels; - af->reverse_endian = 0; } /** @@ -162,9 +163,10 @@ 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: return true; case SAMPLE_FORMAT_UNDEFINED: @@ -180,13 +182,14 @@ audio_valid_sample_format(enum sample_format format) static inline bool audio_valid_channel_count(unsigned channels) { - return channels >= 1 && channels <= 8; + 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) && @@ -198,6 +201,7 @@ static inline bool audio_format_valid(const struct audio_format *af) * 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 || @@ -212,58 +216,54 @@ static inline bool audio_format_equals(const struct audio_format *a, { return a->sample_rate == b->sample_rate && a->format == b->format && - a->channels == b->channels && - a->reverse_endian == b->reverse_endian; + a->channels == b->channels; } -static inline void +void audio_format_mask_apply(struct audio_format *af, - const struct audio_format *mask) -{ - assert(audio_format_valid(af)); - assert(audio_format_mask_valid(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; - - assert(audio_format_valid(af)); -} + const struct audio_format *mask); -/** - * Returns the size of each (mono) sample in bytes. - */ -static inline unsigned audio_format_sample_size(const struct audio_format *af) +G_GNUC_CONST +static inline unsigned +sample_format_size(enum sample_format format) { - switch (af->format) { + switch (format) { case SAMPLE_FORMAT_S8: return 1; case SAMPLE_FORMAT_S16: return 2; - case SAMPLE_FORMAT_S24: - return 3; - 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: - break; + 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) { @@ -274,6 +274,7 @@ audio_format_frame_size(const struct audio_format *af) * 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); @@ -286,6 +287,7 @@ static inline double audio_format_time_to_size(const struct audio_format *af) * @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); @@ -297,6 +299,7 @@ sample_format_to_string(enum sample_format format); * @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); |