audio_format: added audio_format_sample_size()

The inline function audio_format_sample_size() calculates how many
bytes each sample consumes.  This function already takes into account
that 24 bit samples are 4 bytes long, not 3.

src/audioOutputs/audioOutput_alsa.c

@@ -289,7 +289,7 @@ configure_hw:
 	if (err < 0)
 		goto error;
 
-	ad->sampleSize = (audioFormat->bits / 8) * audioFormat->channels;
+	ad->sampleSize = audio_format_sample_size(audioFormat) * audioFormat->channels;
 
 	audioOutput->open = 1;
 

src/audioOutputs/audioOutput_osx.c

@@ -268,7 +268,7 @@ static int osx_openDevice(struct audio_output *audioOutput)
 #endif
 
 	streamDesc.mBytesPerPacket =
-	    audioFormat->channels * audioFormat->bits / 8;
+		audioFormat->channels * audio_format_sample_size(audioFormat);
 	streamDesc.mFramesPerPacket = 1;
 	streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
 	streamDesc.mChannelsPerFrame = audioFormat->channels;

src/audioOutputs/audioOutput_shout_mp3.c

@@ -141,7 +141,7 @@ static int shout_mp3_encoder_encode(struct shout_data *sd,
 	float (*lamebuf)[2];
 	struct shout_buffer *buf = &(sd->buf);
 	unsigned int samples;
-	int bytes = sd->audio_format.bits / 8;
+	int bytes = audio_format_sample_size(&sd->audio_format);
 	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
 	int bytes_out;
 

src/audioOutputs/audioOutput_shout_ogg.c

@@ -257,7 +257,7 @@ static int shout_ogg_encoder_encode(struct shout_data *sd,
 	int j;
 	float **vorbbuf;
 	unsigned int samples;
-	int bytes = sd->audio_format.bits / 8;
+	int bytes = audio_format_sample_size(&sd->audio_format);
 	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
 
 	samples = size / (bytes * sd->audio_format.channels);

src/audio_format.h

@@ -47,14 +47,27 @@ static inline int audio_format_equals(const struct audio_format *a,
 		a->channels == b->channels;
 }
 
+/**
+ * Returns the size of each (mono) sample in bytes.
+ */
+static inline unsigned audio_format_sample_size(const struct audio_format *af)
+{
+	if (af->bits <= 8)
+		return 1;
+	else if (af->bits <= 16)
+		return 2;
+	else
+		return 4;
+}
+
 static inline double audio_format_time_to_size(const struct audio_format *af)
 {
-	return af->sampleRate * af->bits * af->channels / 8.0;
+	return af->sampleRate * af->channels * audio_format_sample_size(af);
 }
 
 static inline double audioFormatSizeToTime(const struct audio_format *af)
 {
-	return 8.0 / af->bits / af->channels / af->sampleRate;
+	return 1.0 / audio_format_time_to_size(af);
 }
 
 #endif

src/inputPlugins/flac_plugin.c

@@ -237,7 +237,8 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
 	FLAC__uint32 samples = frame->header.blocksize;
 	unsigned int c_samp;
 	const unsigned int num_channels = frame->header.channels;
-	const unsigned int bytes_per_sample = (data->audio_format.bits / 8);
+	const unsigned int bytes_per_sample =
+		audio_format_sample_size(&data->audio_format);
 	const unsigned int bytes_per_channel =
 		bytes_per_sample * frame->header.channels;
 	const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;

src/inputPlugins/oggflac_plugin.c

@@ -160,7 +160,7 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(mpd_unused const
 	FLAC__uint16 u16;
 	unsigned char *uc;
 	unsigned int c_samp, c_chan;
-	int i;
+	unsigned int i;
 	float timeChange;
 
 	timeChange = ((float)samples) / frame->header.sample_rate;
@@ -183,7 +183,7 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(mpd_unused const
 		     c_chan++) {
 			u16 = buf[c_chan][c_samp];
 			uc = (unsigned char *)&u16;
-			for (i = 0; i < (data->audio_format.bits / 8); i++) {
+			for (i = 0; i < audio_format_sample_size(&data->audio_format); i++) {
 				if (data->chunk_length >= FLAC_CHUNK_SIZE) {
 					if (flacSendChunk(data) < 0) {
 						return