Add audio_format_init() function

It makes no difference right now, but we're about to add an endianness
flag and will want to make sure it's correctly initialised every time.

15 files changed, 39 insertions, 53 deletions

diff --git a/src/audio_format.h b/src/audio_format.h
index 64087d07..e325c1b3 100644
--- a/src/audio_format.h
+++ b/src/audio_format.h
@@ -36,6 +36,15 @@ static inline void audio_format_clear(struct audio_format *af)
 	af->channels = 0;
 }
 
+static inline void audio_format_init(struct audio_format *af,
+				     uint32_t sample_rate,
+				     uint8_t bits, uint8_t channels)
+{
+	af->sample_rate = sample_rate;
+	af->bits = bits;
+	af->channels = channels;
+}
+
 static inline bool audio_format_defined(const struct audio_format *af)
 {
 	return af->sample_rate != 0;
diff --git a/src/audio_parser.c b/src/audio_parser.c
index 906b0f81..d29f5f44 100644
--- a/src/audio_parser.c
+++ b/src/audio_parser.c
@@ -41,6 +41,8 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error)
 {
 	char *endptr;
 	unsigned long value;
+	uint32_t rate;
+	uint8_t bits, channels;
 
 	audio_format_clear(dest);
 
@@ -61,7 +63,7 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error)
 		return false;
 	}
 
-	dest->sample_rate = value;
+	rate = value;
 
 	/* parse sample format */
 
@@ -81,7 +83,7 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error)
 		return false;
 	}
 
-	dest->bits = value;
+	bits = value;
 
 	/* parse channel count */
 
@@ -93,7 +95,9 @@ audio_format_parse(struct audio_format *dest, const char *src, GError **error)
 		return false;
 	}
 
-	dest->channels = value;
+	channels = value;
+
+	audio_format_init(dest, rate, bits, channels);
 
 	return true;
 }
diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index 713dfe9b..7b345385 100644
--- a/src/decoder/_flac_common.c
+++ b/src/decoder/_flac_common.c
@@ -195,9 +195,8 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		data->audio_format.bits = (int8_t)si->bits_per_sample;
-		data->audio_format.sample_rate = si->sample_rate;
-		data->audio_format.channels = (int8_t)si->channels;
+		audio_format_init(&data->audio_format, si->sample_rate,
+				  si->bits_per_sample, si->channels);
 		data->total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/decoder/audiofile_plugin.c b/src/decoder/audiofile_plugin.c
index f66d90dc..b4959f6c 100644
--- a/src/decoder/audiofile_plugin.c
+++ b/src/decoder/audiofile_plugin.c
@@ -136,11 +136,9 @@ audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, bits);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	audio_format.bits = (uint8_t)bits;
-	audio_format.sample_rate =
-	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
-	audio_format.channels =
-	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+
+	audio_format_init(&audio_format, afGetRate(af_fp, AF_DEFAULT_TRACK),
+			  bits, afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK)); 
 
 	if (!audio_format_valid(&audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/faad_plugin.c b/src/decoder/faad_plugin.c
index d0537dd5..1b8b2b78 100644
--- a/src/decoder/faad_plugin.c
+++ b/src/decoder/faad_plugin.c
@@ -262,11 +262,7 @@ faad_decoder_init(faacDecHandle decoder, struct decoder_buffer *buffer,
 
 	decoder_buffer_consume(buffer, nbytes);
 
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
+	audio_format_init(audio_format, sample_rate, 16, channels);
 
 	return true;
 }
diff --git a/src/decoder/ffmpeg_plugin.c b/src/decoder/ffmpeg_plugin.c
index 03c46a73..f6003d2f 100644
--- a/src/decoder/ffmpeg_plugin.c
+++ b/src/decoder/ffmpeg_plugin.c
@@ -267,6 +267,7 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	struct audio_format audio_format;
 	enum decoder_command cmd;
 	int total_time;
+	uint8_t bits;
 
 	total_time = 0;
 
@@ -275,13 +276,13 @@ ffmpeg_decode_internal(struct ffmpeg_context *ctx)
 	}
 
 #if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
-	audio_format.bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
+	bits = (uint8_t) av_get_bits_per_sample_format(codec_context->sample_fmt);
 #else
 	/* XXX fixme 16-bit for older ffmpeg (13 Aug 2007) */
-	audio_format.bits = (uint8_t) 16;
+	bits = (uint8_t) 16;
 #endif
-	audio_format.sample_rate = (unsigned int)codec_context->sample_rate;
-	audio_format.channels = codec_context->channels;
+	audio_format_init(&audio_format, codec_context->sample_rate, bits,
+			  codec_context->channels);
 
 	if (!audio_format_valid(&audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mad_plugin.c b/src/decoder/mad_plugin.c
index c6b9d32d..85f4506d 100644
--- a/src/decoder/mad_plugin.c
+++ b/src/decoder/mad_plugin.c
@@ -1148,13 +1148,6 @@ mp3_read(struct mp3_data *data, struct replay_gain_info **replay_gain_info_r)
 	return ret != DECODE_BREAK;
 }
 
-static void mp3_audio_format(struct mp3_data *data, struct audio_format *af)
-{
-	af->bits = 24;
-	af->sample_rate = (data->frame).header.samplerate;
-	af->channels = MAD_NCHANNELS(&(data->frame).header);
-}
-
 static void
 mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 {
@@ -1170,7 +1163,8 @@ mp3_decode(struct decoder *decoder, struct input_stream *input_stream)
 		return;
 	}
 
-	mp3_audio_format(&data, &audio_format);
+	audio_format_init(&audio_format, data.frame.header.samplerate, 24,
+			  MAD_NCHANNELS(&data.frame.header));
 
 	decoder_initialized(decoder, &audio_format,
 			    data.input_stream->seekable, data.total_time);
diff --git a/src/decoder/mikmod_plugin.c b/src/decoder/mikmod_plugin.c
index 065c3431..e7b7bfb0 100644
--- a/src/decoder/mikmod_plugin.c
+++ b/src/decoder/mikmod_plugin.c
@@ -175,9 +175,7 @@ mod_decode(struct decoder *decoder, const char *path)
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 44100, 16, 2);
 
 	secPerByte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/modplug_plugin.c b/src/decoder/modplug_plugin.c
index 31f0a47c..6c375e6a 100644
--- a/src/decoder/modplug_plugin.c
+++ b/src/decoder/modplug_plugin.c
@@ -121,9 +121,7 @@ mod_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	audio_format.bits = 16;
-	audio_format.sample_rate = 44100;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 44100, 16, 2);
 
 	sec_perbyte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
diff --git a/src/decoder/mp4ff_plugin.c b/src/decoder/mp4ff_plugin.c
index cf938290..d2c63f98 100644
--- a/src/decoder/mp4ff_plugin.c
+++ b/src/decoder/mp4ff_plugin.c
@@ -131,11 +131,7 @@ mp4_faad_new(mp4ff_t *mp4fh, int *track_r, struct audio_format *audio_format)
 	}
 
 	*track_r = track;
-	*audio_format = (struct audio_format){
-		.bits = 16,
-		.channels = channels,
-		.sample_rate = sample_rate,
-	};
+	audio_format_init(audio_format, sample_rate, 16, channels);
 
 	if (!audio_format_valid(audio_format)) {
 		g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/mpcdec_plugin.c b/src/decoder/mpcdec_plugin.c
index 26349f93..a684da10 100644
--- a/src/decoder/mpcdec_plugin.c
+++ b/src/decoder/mpcdec_plugin.c
@@ -193,9 +193,7 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_demux_get_info(demux, &info);
 #endif
 
-	audio_format.bits = 24;
-	audio_format.channels = info.channels;
-	audio_format.sample_rate = info.sample_freq;
+	audio_format_init(&audio_format, info.sample_freq, 24, info.channels);
 
 	if (!audio_format_valid(&audio_format)) {
 #ifndef MPC_IS_OLD_API
diff --git a/src/decoder/sidplay_plugin.cxx b/src/decoder/sidplay_plugin.cxx
index c62e6b4b..54ab746e 100644
--- a/src/decoder/sidplay_plugin.cxx
+++ b/src/decoder/sidplay_plugin.cxx
@@ -103,9 +103,7 @@ sidplay_file_decode(struct decoder *decoder, const char *path_fs)
 	/* initialize the MPD decoder */
 
 	struct audio_format audio_format;
-	audio_format.sample_rate = 48000;
-	audio_format.bits = 16;
-	audio_format.channels = 2;
+	audio_format_init(&audio_format, 48000, 16, 2);
 
 	decoder_initialized(decoder, &audio_format, false, -1);
 
diff --git a/src/decoder/sndfile_decoder_plugin.c b/src/decoder/sndfile_decoder_plugin.c
index 0c5d2f06..4cc64459 100644
--- a/src/decoder/sndfile_decoder_plugin.c
+++ b/src/decoder/sndfile_decoder_plugin.c
@@ -124,12 +124,10 @@ sndfile_stream_decode(struct decoder *decoder, struct input_stream *is)
 		return;
 	}
 
-	audio_format.sample_rate = info.samplerate;
 	/* for now, always read 32 bit samples.  Later, we could lower
 	   MPD's CPU usage by reading 16 bit samples with
 	   sf_readf_short() on low-quality source files. */
-	audio_format.bits = 32;
-	audio_format.channels = info.channels;
+	audio_format_init(&audio_format, info.samplerate, 32, info.channels);
 
 	if (!audio_format_valid(&audio_format)) {
 		g_warning("invalid audio format");
diff --git a/src/decoder/vorbis_plugin.c b/src/decoder/vorbis_plugin.c
index d4f81e91..bab1d57e 100644
--- a/src/decoder/vorbis_plugin.c
+++ b/src/decoder/vorbis_plugin.c
@@ -324,8 +324,7 @@ vorbis_stream_decode(struct decoder *decoder,
 			vorbis_info *vi = ov_info(&vf, -1);
 			struct replay_gain_info *new_rgi;
 
-			audio_format.channels = vi->channels;
-			audio_format.sample_rate = vi->rate;
+			audio_format_init(&audio_format, vi->rate, 16, vi->channels);
 
 			if (!audio_format_valid(&audio_format)) {
 				g_warning("Invalid audio format: %u:%u:%u\n",
diff --git a/src/decoder/wavpack_plugin.c b/src/decoder/wavpack_plugin.c
index 821536fb..f3d70114 100644
--- a/src/decoder/wavpack_plugin.c
+++ b/src/decoder/wavpack_plugin.c
@@ -145,9 +145,9 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
 	int bytes_per_sample, output_sample_size;
 	int position;
 
-	audio_format.sample_rate = WavpackGetSampleRate(wpc);
-	audio_format.channels = WavpackGetReducedChannels(wpc);
-	audio_format.bits = WavpackGetBitsPerSample(wpc);
+	audio_format_init(&audio_format, WavpackGetSampleRate(wpc),
+			  WavpackGetBitsPerSample(wpc), 
+			  WavpackGetReducedChannels(wpc));
 
 	/* round bitwidth to 8-bit units */
 	audio_format.bits = (audio_format.bits + 7) & (~7);