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/audio.c

@@ -17,6 +17,7 @@
  */
 
 #include "audio.h"
+#include "audio_format.h"
 #include "audioOutput.h"
 #include "log.h"
 #include "path.h"
@@ -27,9 +28,9 @@
 #define AUDIO_DEVICE_STATE_LEN	(sizeof(AUDIO_DEVICE_STATE)-1)
 #define AUDIO_BUFFER_SIZE	2*MPD_PATH_MAX
 
-static AudioFormat audio_format;
+static struct audio_format audio_format;
 
-static AudioFormat *audio_configFormat;
+static struct audio_format *audio_configFormat;
 
 static AudioOutput *audioOutputArray;
 static unsigned int audioOutputArraySize;
@@ -63,15 +64,15 @@ static unsigned int audio_output_count(void)
 	return nr;
 }
 
-void copyAudioFormat(AudioFormat * dest, const AudioFormat * src)
+void copyAudioFormat(struct audio_format *dest, const struct audio_format *src)
 {
 	if (!src)
 		return;
 
-	memcpy(dest, src, sizeof(AudioFormat));
+	memcpy(dest, src, sizeof(*dest));
 }
 
-int cmpAudioFormat(const AudioFormat * f1, const AudioFormat * f2)
+int cmpAudioFormat(const struct audio_format *f1, const struct audio_format *f2)
 {
 	if (f1 && f2 && (f1->sampleRate == f2->sampleRate) &&
 	    (f1->bits == f2->bits) && (f1->channels == f2->channels))
@@ -141,8 +142,8 @@ void initAudioDriver(void)
 	}
 }
 
-void getOutputAudioFormat(const AudioFormat * inAudioFormat,
-			  AudioFormat * outAudioFormat)
+void getOutputAudioFormat(const struct audio_format *inAudioFormat,
+			  struct audio_format *outAudioFormat)
 {
 	if (audio_configFormat) {
 		copyAudioFormat(outAudioFormat, audio_configFormat);
@@ -157,7 +158,7 @@ void initAudioConfig(void)
 	if (NULL == param || NULL == param->value)
 		return;
 
-	audio_configFormat = xmalloc(sizeof(AudioFormat));
+	audio_configFormat = xmalloc(sizeof(*audio_configFormat));
 
 	if (0 != parseAudioConfig(audio_configFormat, param->value)) {
 		FATAL("error parsing \"%s\" at line %i\n",
@@ -165,11 +166,11 @@ void initAudioConfig(void)
 	}
 }
 
-int parseAudioConfig(AudioFormat * audioFormat, char *conf)
+int parseAudioConfig(struct audio_format *audioFormat, char *conf)
 {
 	char *test;
 
-	memset(audioFormat, 0, sizeof(AudioFormat));
+	memset(audioFormat, 0, sizeof(*audioFormat));
 
 	audioFormat->sampleRate = strtol(conf, &test, 10);
 
@@ -251,7 +252,7 @@ void finishAudioDriver(void)
 	audioOutputArraySize = 0;
 }
 
-int isCurrentAudioFormat(const AudioFormat * audioFormat)
+int isCurrentAudioFormat(const struct audio_format *audioFormat)
 {
 	if (!audioFormat)
 		return 1;
@@ -320,7 +321,7 @@ static int flushAudioBuffer(void)
 	return ret;
 }
 
-int openAudioDevice(const AudioFormat * audioFormat)
+int openAudioDevice(const struct audio_format *audioFormat)
 {
 	int ret = -1;
 	unsigned int i;

src/audio.h

@@ -20,21 +20,21 @@
 #define AUDIO_H
 
 #include "os_compat.h"
-#include "audio_format.h"
 
 #define AUDIO_AO_DRIVER_DEFAULT	"default"
 
+struct audio_format;
 struct tag;
 struct client;
 
-void copyAudioFormat(AudioFormat * dest, const AudioFormat * src);
+void copyAudioFormat(struct audio_format *dest, const struct audio_format *src);
 
-int cmpAudioFormat(const AudioFormat * dest, const AudioFormat * src);
+int cmpAudioFormat(const struct audio_format *dest, const struct audio_format *src);
 
-void getOutputAudioFormat(const AudioFormat * inFormat,
-			  AudioFormat * outFormat);
+void getOutputAudioFormat(const struct audio_format *inFormat,
+			  struct audio_format *outFormat);
 
-int parseAudioConfig(AudioFormat * audioFormat, char *conf);
+int parseAudioConfig(struct audio_format *audioFormat, char *conf);
 
 /* make sure initPlayerData is called before this function!! */
 void initAudioConfig(void);
@@ -45,7 +45,7 @@ void initAudioDriver(void);
 
 void finishAudioDriver(void);
 
-int openAudioDevice(const AudioFormat * audioFormat);
+int openAudioDevice(const struct audio_format *audioFormat);
 
 int playAudio(const char *playChunk, size_t size);
 
@@ -55,7 +55,7 @@ void closeAudioDevice(void);
 
 int isAudioDeviceOpen(void);
 
-int isCurrentAudioFormat(const AudioFormat * audioFormat);
+int isCurrentAudioFormat(const struct audio_format *audioFormat);
 
 void sendMetadataToAudioDevice(const struct tag *tag);
 

src/audioOutput.c

@@ -130,9 +130,9 @@ int initAudioOutput(AudioOutput *ao, ConfigParam * param)
 	ao->convBuffer = NULL;
 	ao->convBufferLen = 0;
 
-	memset(&ao->inAudioFormat, 0, sizeof(AudioFormat));
-	memset(&ao->outAudioFormat, 0, sizeof(AudioFormat));
-	memset(&ao->reqAudioFormat, 0, sizeof(AudioFormat));
+	memset(&ao->inAudioFormat, 0, sizeof(ao->inAudioFormat));
+	memset(&ao->outAudioFormat, 0, sizeof(ao->outAudioFormat));
+	memset(&ao->reqAudioFormat, 0, sizeof(ao->reqAudioFormat));
 	memset(&ao->convState, 0, sizeof(ConvState));
 
 	if (format) {
@@ -152,7 +152,7 @@ int initAudioOutput(AudioOutput *ao, ConfigParam * param)
 }
 
 int openAudioOutput(AudioOutput * audioOutput,
-		    const AudioFormat * audioFormat)
+		    const struct audio_format *audioFormat)
 {
 	int ret = 0;
 

src/audioOutput.h

@@ -65,9 +65,9 @@ struct _AudioOutput {
 	AudioOutputSendMetadataFunc sendMetdataFunc;
 
 	int convertAudioFormat;
-	AudioFormat inAudioFormat;
-	AudioFormat outAudioFormat;
-	AudioFormat reqAudioFormat;
+	struct audio_format inAudioFormat;
+	struct audio_format outAudioFormat;
+	struct audio_format reqAudioFormat;
 	ConvState convState;
 	char *convBuffer;
 	size_t convBufferLen;
@@ -97,7 +97,7 @@ void unloadAudioOutputPlugin(AudioOutputPlugin * audioOutputPlugin);
 
 int initAudioOutput(AudioOutput *, ConfigParam * param);
 int openAudioOutput(AudioOutput * audioOutput,
-		    const AudioFormat * audioFormat);
+		    const struct audio_format *audioFormat);
 int playAudioOutput(AudioOutput * audioOutput,
 		    const char *playChunk, size_t size);
 void dropBufferedAudioOutput(AudioOutput * audioOutput);

src/audioOutputs/audioOutput_alsa.c

@@ -123,7 +123,7 @@ static int alsa_testDefault(void)
 static int alsa_openDevice(AudioOutput * audioOutput)
 {
 	AlsaData *ad = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	struct audio_format *audioFormat = &audioOutput->outAudioFormat;
 	snd_pcm_format_t bitformat;
 	snd_pcm_hw_params_t *hwparams;
 	snd_pcm_sw_params_t *swparams;

src/audioOutputs/audioOutput_jack.c

@@ -118,7 +118,7 @@ static void jack_finishDriver(AudioOutput *audioOutput)
 static int srate(mpd_unused jack_nframes_t rate, void *data)
 {
 	JackData *jd = (JackData *) ((AudioOutput*) data)->data;
- 	AudioFormat *audioFormat = &(((AudioOutput*) data)->outAudioFormat);
+ 	struct audio_format *audioFormat = &(((AudioOutput*) data)->outAudioFormat);
 
  	audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client);
 
@@ -183,7 +183,7 @@ static void shutdown_callback(void *arg)
 static void set_audioformat(AudioOutput *audioOutput)
 {
 	JackData *jd = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	struct audio_format *audioFormat = &audioOutput->outAudioFormat;
 
 	audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client);
 	DEBUG("samplerate = %d\n", audioFormat->sampleRate);

src/audioOutputs/audioOutput_oss.c

@@ -484,7 +484,7 @@ static int oss_openDevice(AudioOutput * audioOutput)
 {
 	int ret;
 	OssData *od = audioOutput->data;
-	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
+	struct audio_format *audioFormat = &audioOutput->outAudioFormat;
 
 	od->channels = (mpd_sint8)audioFormat->channels;
 	od->sampleRate = audioFormat->sampleRate;

src/audioOutputs/audioOutput_pulse.c

@@ -112,7 +112,7 @@ static int pulse_testDefault(void)
 static int pulse_openDevice(AudioOutput * audioOutput)
 {
 	PulseData *pd;
-	AudioFormat *audioFormat;
+	struct audio_format *audioFormat;
 	pa_sample_spec ss;
 	time_t t;
 	int error;

src/audioOutputs/audioOutput_shout.c

@@ -66,7 +66,7 @@ typedef struct _ShoutData {
 	Timer *timer;
 
 	/* just a pointer to audioOutput->outAudioFormat */
-	AudioFormat *audioFormat;
+	struct audio_format *audioFormat;
 } ShoutData;
 
 static ShoutData *newShoutData(void)

src/audio_format.h

@@ -21,18 +21,18 @@
 
 #include "mpd_types.h"
 
-typedef struct _AudioFormat {
+struct audio_format {
 	mpd_sint8 channels;
 	mpd_uint32 sampleRate;
 	mpd_sint8 bits;
-} AudioFormat;
+};
 
-static inline double audio_format_time_to_size(const AudioFormat * af)
+static inline double audio_format_time_to_size(const struct audio_format *af)
 {
 	return af->sampleRate * af->bits * af->channels / 8.0;
 }
 
-static inline double audioFormatSizeToTime(const AudioFormat * af)
+static inline double audioFormatSizeToTime(const struct audio_format *af)
 {
 	return 8.0 / af->bits / af->channels / af->sampleRate;
 }

src/crossfade.c

@@ -22,7 +22,7 @@
 #include "pcm_utils.h"
 
 unsigned cross_fade_calc(float duration, float total_time,
-			 const AudioFormat * af,
+			 const struct audio_format *af,
 			 unsigned max_chunks)
 {
 	unsigned int chunks;
@@ -46,7 +46,7 @@ unsigned cross_fade_calc(float duration, float total_time,
 }
 
 void cross_fade_apply(ob_chunk * a, const ob_chunk * b,
-		      const AudioFormat * format,
+		      const struct audio_format *format,
 		      unsigned int current_chunk, unsigned int num_chunks)
 {
 	assert(current_chunk <= num_chunks);

src/crossfade.h

@@ -20,15 +20,16 @@
 #ifndef CROSSFADE_H
 #define CROSSFADE_H
 
-#include "audio_format.h"
 #include "outputBuffer.h"
 
+struct audio_format;
+
 unsigned cross_fade_calc(float duration, float total_time,
-			 const AudioFormat * af,
+			 const struct audio_format *af,
 			 unsigned max_chunks);
 
 void cross_fade_apply(ob_chunk * a, const ob_chunk * b,
-		      const AudioFormat * format,
+		      const struct audio_format *format,
 		      unsigned int current_chunk, unsigned int num_chunks);
 
 #endif

src/decoder_api.c

@@ -39,7 +39,7 @@ void decoder_plugin_unregister(struct decoder_plugin *plugin)
 }
 
 void decoder_initialized(struct decoder * decoder,
-			 const AudioFormat * audio_format,
+			 const struct audio_format *audio_format,
 			 float total_time)
 {
 	assert(dc.state == DECODE_STATE_START);

src/decoder_api.h

@@ -30,6 +30,7 @@
 #include "replayGain.h"
 #include "tag.h"
 #include "tag_id3.h"
+#include "audio_format.h"
 #include "playerData.h"
 
 
@@ -112,7 +113,7 @@ struct decoder;
  * that it has read the song's meta data.
  */
 void decoder_initialized(struct decoder * decoder,
-			 const AudioFormat * audio_format,
+			 const struct audio_format *audio_format,
 			 float total_time);
 
 const char *decoder_get_url(struct decoder * decoder, char * buffer);

src/decoder_control.h

@@ -48,7 +48,7 @@ struct decoder_control {
 	volatile mpd_sint8 seekError;
 	volatile mpd_sint8 seekable;
 	volatile double seekWhere;
-	AudioFormat audioFormat;
+	struct audio_format audioFormat;
 	Song *current_song;
 	Song *volatile next_song;
 	volatile float totalTime;

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;

src/normalize.c

@@ -19,6 +19,7 @@
 #include "normalize.h"
 #include "compress.h"
 #include "conf.h"
+#include "audio_format.h"
 
 #define DEFAULT_VOLUME_NORMALIZATION 0
 
@@ -39,7 +40,8 @@ void finishNormalization(void)
 	if (normalizationEnabled) CompressFree();
 }
 
-void normalizeData(char *buffer, int bufferSize, const AudioFormat *format)
+void normalizeData(char *buffer, int bufferSize,
+		   const struct audio_format *format)
 {
 	if ((format->bits != 16) || (format->channels != 2)) return;
 

src/normalize.h

@@ -19,7 +19,7 @@
 #ifndef NORMALIZE_H
 #define NORMALIZE_H
 
-#include "audio_format.h"
+struct audio_format;
 
 extern int normalizationEnabled;
 
@@ -27,6 +27,7 @@ void initNormalization(void);
 
 void finishNormalization(void);
 
-void normalizeData(char *buffer, int bufferSize, const AudioFormat *format);
+void normalizeData(char *buffer, int bufferSize,
+		   const struct audio_format *format);
 
 #endif /* !NORMALIZE_H */

src/outputBuffer.h

@@ -51,7 +51,7 @@ struct output_buffer {
 	    the buffer becomes non-empty */
 	int lazy;
 
-	AudioFormat audioFormat;
+	struct audio_format audioFormat;
 
 	Notify *notify;
 };

src/pcm_utils.c

@@ -22,9 +22,11 @@
 #include "log.h"
 #include "utils.h"
 #include "conf.h"
+#include "audio_format.h"
 #include "os_compat.h"
 
-void pcm_volumeChange(char *buffer, int bufferSize, const AudioFormat * format,
+void pcm_volumeChange(char *buffer, int bufferSize,
+                      const struct audio_format *format,
                       int volume)
 {
 	mpd_sint32 temp32;
@@ -76,7 +78,7 @@ void pcm_volumeChange(char *buffer, int bufferSize, const AudioFormat * format,
 
 static void pcm_add(char *buffer1, const char *buffer2, size_t bufferSize1,
                     size_t bufferSize2, int vol1, int vol2,
-                    const AudioFormat * format)
+                    const struct audio_format *format)
 {
 	mpd_sint32 temp32;
 	mpd_sint8 *buffer8_1 = (mpd_sint8 *) buffer1;
@@ -130,7 +132,8 @@ static void pcm_add(char *buffer1, const char *buffer2, size_t bufferSize1,
 }
 
 void pcm_mix(char *buffer1, const char *buffer2, size_t bufferSize1,
-             size_t bufferSize2, const AudioFormat * format, float portion1)
+             size_t bufferSize2, const struct audio_format *format,
+             float portion1)
 {
 	int vol1;
 	float s = sin(M_PI_2 * portion1);
@@ -392,9 +395,9 @@ static const char *pcm_convertTo16bit(mpd_sint8 bits, const char *inBuffer,
 }
 
 /* outFormat bits must be 16 and channels must be 1 or 2! */
-size_t pcm_convertAudioFormat(const AudioFormat * inFormat,
+size_t pcm_convertAudioFormat(const struct audio_format *inFormat,
 			      const char *inBuffer, size_t inSize,
-			      const AudioFormat * outFormat,
+			      const struct audio_format *outFormat,
                               char *outBuffer, ConvState *convState)
 {
 	const char *buf;
@@ -430,8 +433,8 @@ size_t pcm_convertAudioFormat(const AudioFormat * inFormat,
 	return len;
 }
 
-size_t pcm_sizeOfConvBuffer(const AudioFormat * inFormat, size_t inSize,
-                            const AudioFormat * outFormat)
+size_t pcm_sizeOfConvBuffer(const struct audio_format *inFormat, size_t inSize,
+                            const struct audio_format *outFormat)
 {
 	const double ratio = (double)outFormat->sampleRate /
 	                     (double)inFormat->sampleRate;

src/pcm_utils.h

@@ -21,13 +21,15 @@
 
 #include "../config.h"
 
-#include "audio_format.h"
+#include "mpd_types.h"
 #include "os_compat.h"
 
 #ifdef HAVE_LIBSAMPLERATE
 #include <samplerate.h>
 #endif
 
+struct audio_format;
+
 typedef struct _ConvState {
 #ifdef HAVE_LIBSAMPLERATE
 	SRC_STATE *state;
@@ -42,17 +44,17 @@ typedef struct _ConvState {
 	int error;
 } ConvState;
 
-void pcm_volumeChange(char *buffer, int bufferSize, const AudioFormat * format,
+void pcm_volumeChange(char *buffer, int bufferSize, const struct audio_format *format,
                       int volume);
 
 void pcm_mix(char *buffer1, const char *buffer2, size_t bufferSize1,
-             size_t bufferSize2, const AudioFormat * format, float portion1);
+             size_t bufferSize2, const struct audio_format *format, float portion1);
 
-size_t pcm_convertAudioFormat(const AudioFormat * inFormat,
+size_t pcm_convertAudioFormat(const struct audio_format *inFormat,
 			      const char *inBuffer, size_t inSize,
-			      const AudioFormat * outFormat,
+			      const struct audio_format *outFormat,
                               char *outBuffer, ConvState *convState);
 
-size_t pcm_sizeOfConvBuffer(const AudioFormat * inFormat, size_t inSize,
-                            const AudioFormat * outFormat);
+size_t pcm_sizeOfConvBuffer(const struct audio_format *inFormat, size_t inSize,
+                            const struct audio_format *outFormat);
 #endif

src/player_thread.c

@@ -158,7 +158,7 @@ static void processDecodeInput(int *pause_r, unsigned int *bbp_r,
 }
 
 static int playChunk(ob_chunk * chunk,
-		     const AudioFormat * format, double sizeToTime)
+		     const struct audio_format *format, double sizeToTime)
 {
 	pc.elapsedTime = chunk->times;
 	pc.bitRate = chunk->bitRate;

src/replayGain.c

@@ -22,6 +22,8 @@
 
 #include "log.h"
 #include "conf.h"
+#include "audio_format.h"
+#include "mpd_types.h"
 #include "os_compat.h"
 
 /* Added 4/14/2004 by AliasMrJones */
@@ -104,7 +106,7 @@ void freeReplayGainInfo(ReplayGainInfo * info)
 }
 
 void doReplayGain(ReplayGainInfo * info, char *buffer, int bufferSize,
-		  const AudioFormat * format)
+		  const struct audio_format *format)
 {
 	mpd_sint16 *buffer16;
 	mpd_sint8 *buffer8;

src/replayGain.h

@@ -20,12 +20,12 @@
 #ifndef REPLAYGAIN_H
 #define REPLAYGAIN_H
 
-#include "audio_format.h"
-
 #define REPLAYGAIN_OFF		0
 #define REPLAYGAIN_TRACK	1
 #define REPLAYGAIN_ALBUM	2
 
+struct audio_format;
+
 extern int replayGainState;
 
 typedef struct _ReplayGainInfo {
@@ -45,6 +45,6 @@ void freeReplayGainInfo(ReplayGainInfo * info);
 void initReplayGainState(void);
 
 void doReplayGain(ReplayGainInfo * info, char *buffer, int bufferSize,
-		  const AudioFormat * format);
+		  const struct audio_format *format);
 
 #endif

src/timer.c

@@ -18,6 +18,7 @@
 
 #include "timer.h"
 #include "utils.h"
+#include "audio_format.h"
 #include "os_compat.h"
 
 static uint64_t now(void)
@@ -29,7 +30,7 @@ static uint64_t now(void)
 	return ((uint64_t)tv.tv_sec * 1000000) + tv.tv_usec;
 }
 
-Timer *timer_new(AudioFormat *af)
+Timer *timer_new(struct audio_format *af)
 {
 	Timer *timer;
 

src/timer.h

@@ -19,16 +19,17 @@
 #ifndef MPD_TIMER_H
 #define MPD_TIMER_H
 
-#include "audio_format.h"
 #include "os_compat.h"
 
+struct audio_format;
+
 typedef struct _Timer {
 	uint64_t time;
 	int started;
 	int rate;
 } Timer;
 
-Timer *timer_new(AudioFormat *af);
+Timer *timer_new(struct audio_format *af);
 
 void timer_free(Timer *timer);