audio_format: converted typedef AudioFormat to struct audio_format

Get rid of CamelCase, and don't use a typedef, so we can
forward-declare it, and unclutter the include dependencies.

src/inputPlugins/_flac_common.h

@@ -143,7 +143,7 @@ typedef struct {
 	size_t chunk_length;
 	float time;
 	unsigned int bitRate;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	float total_time;
 	FLAC__uint64 position;
 	struct decoder *decoder;

src/inputPlugins/aac_plugin.c

@@ -310,7 +310,7 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
 	long bread;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	uint32_t sampleRate;
 	unsigned char channels;
 	unsigned int sampleCount;
@@ -444,7 +444,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
 	long bread;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	uint32_t sampleRate;
 	unsigned char channels;
 	unsigned int sampleCount;

src/inputPlugins/audiofile_plugin.c

@@ -48,7 +48,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	int fs, frame_count;
 	AFfilehandle af_fp;
 	int bits;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	float total_time;
 	mpd_uint16 bitRate;
 	struct stat st;

src/inputPlugins/mod_plugin.c

@@ -171,7 +171,7 @@ static void mod_close(mod_Data * data)
 static int mod_decode(struct decoder * decoder, char *path)
 {
 	mod_Data *data;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	float total_time = 0.0;
 	int ret;
 	float secPerByte;

src/inputPlugins/mp3_plugin.c

@@ -1021,7 +1021,7 @@ mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
 }
 
 static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
-					     AudioFormat * af)
+					     struct audio_format * af)
 {
 	af->bits = 16;
 	af->sampleRate = (data->frame).header.samplerate;
@@ -1033,7 +1033,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 	mp3DecodeData data;
 	struct tag *tag = NULL;
 	ReplayGainInfo *replayGainInfo = NULL;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 
 	if (openMp3FromInputStream(inStream, &data, decoder,
 				   &tag, &replayGainInfo) < 0) {

src/inputPlugins/mp4_plugin.c

@@ -88,7 +88,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	unsigned char *mp4Buffer;
 	unsigned int mp4BufferSize;
 	uint32_t sampleRate;

src/inputPlugins/mpc_plugin.c

@@ -102,7 +102,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	mpc_decoder decoder;
 	mpc_reader reader;
 	mpc_streaminfo info;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 
 	MpcCallbackData data;
 

src/inputPlugins/oggvorbis_plugin.c

@@ -210,7 +210,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 	OggVorbis_File vf;
 	ov_callbacks callbacks;
 	OggCallbackData data;
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	int current_section;
 	int prev_section = -1;
 	long ret;

src/inputPlugins/wavpack_plugin.c

@@ -131,7 +131,7 @@ static void wavpack_decode(struct decoder * decoder,
                            WavpackContext *wpc, int canseek,
                            ReplayGainInfo *replayGainInfo)
 {
-	AudioFormat audio_format;
+	struct audio_format audio_format;
 	void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
 	char chunk[CHUNK_SIZE];
 	float file_time;