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.

13 files changed, 23 insertions, 50 deletions

diff --git a/src/decoder/_flac_common.c b/src/decoder/_flac_common.c
index 713dfe9b2..7b3453854 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 f66d90dc1..b4959f6c2 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 d0537dd5b..1b8b2b784 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 03c46a732..f6003d2f3 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 c6b9d32d3..85f4506d2 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 065c34319..e7b7bfb03 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 31f0a47c2..6c375e6a0 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 cf9382904..d2c63f983 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 26349f93a..a684da104 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 c62e6b4b6..54ab746e2 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 0c5d2f063..4cc64459f 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 d4f81e91f..bab1d57ec 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 821536fb5..f3d701144 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);