added audio_format parameter to decoder_initialized()

dc->audioFormat is set once by the decoder plugins before invoking
decoder_initialized(); hide dc->audioFormat and let the decoder pass
an AudioFormat pointer to decoder_initialized().

14 files changed, 88 insertions, 83 deletions

diff --git a/src/decoder_api.c b/src/decoder_api.c
index 99f7293bb..9583c7493 100644
--- a/src/decoder_api.c
+++ b/src/decoder_api.c
@@ -24,10 +24,17 @@
 #include "playerData.h"
 #include "gcc.h"
 
-void decoder_initialized(mpd_unused struct decoder * decoder)
+void decoder_initialized(mpd_unused struct decoder * decoder,
+			 const AudioFormat * audio_format)
 {
 	assert(dc.state == DECODE_STATE_START);
 
+	if (audio_format != NULL) {
+		dc.audioFormat = *audio_format;
+		getOutputAudioFormat(audio_format,
+				     &(ob.audioFormat));
+	}
+
 	dc.state = DECODE_STATE_DECODE;
 	notify_signal(&pc.notify);
 }
diff --git a/src/decoder_api.h b/src/decoder_api.h
index a4fd7fc97..eb2ca3887 100644
--- a/src/decoder_api.h
+++ b/src/decoder_api.h
@@ -40,7 +40,8 @@ struct decoder;
  * Notify the player thread that it has finished initialization and
  * that it has read the song's meta data.
  */
-void decoder_initialized(struct decoder * decoder);
+void decoder_initialized(struct decoder * decoder,
+			 const AudioFormat * audio_format);
 
 /**
  * This function is called by the decoder plugin when it has
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index e1017e596..c48f43da9 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -162,11 +162,10 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		dc.audioFormat.bits = (mpd_sint8)si->bits_per_sample;
-		dc.audioFormat.sampleRate = si->sample_rate;
-		dc.audioFormat.channels = (mpd_sint8)si->channels;
+		data->audio_format.bits = (mpd_sint8)si->bits_per_sample;
+		data->audio_format.sampleRate = si->sample_rate;
+		data->audio_format.channels = (mpd_sint8)si->channels;
 		dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
-		getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
 		flacParseReplayGain(block, data);
diff --git a/src/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
index 9eb916b8c..e87ae9307 100644
--- a/src/inputPlugins/_flac_common.h
+++ b/src/inputPlugins/_flac_common.h
@@ -143,6 +143,7 @@ typedef struct {
 	size_t chunk_length;
 	float time;
 	unsigned int bitRate;
+	AudioFormat audio_format;
 	FLAC__uint64 position;
 	struct decoder *decoder;
 	InputStream *inStream;
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index 8765dc21c..4fa54c646 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -286,6 +286,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
 	long bread;
+	AudioFormat audio_format;
 	uint32_t sampleRate;
 	unsigned char channels;
 	int eof = 0;
@@ -335,7 +336,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 		return -1;
 	}
 
-	dc.audioFormat.bits = 16;
+	audio_format.bits = 16;
 
 	dc.totalTime = totalTime;
 
@@ -369,11 +370,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 #endif
 
 		if (dc.state != DECODE_STATE_DECODE) {
-			dc.audioFormat.channels = frameInfo.channels;
-			dc.audioFormat.sampleRate = sampleRate;
-			getOutputAudioFormat(&(dc.audioFormat),
-					     &(ob.audioFormat));
-			decoder_initialized(mpd_decoder);
+			audio_format.channels = frameInfo.channels;
+			audio_format.sampleRate = sampleRate;
+			decoder_initialized(mpd_decoder, &audio_format);
 		}
 
 		advanceAacBuffer(&b, frameInfo.bytesconsumed);
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 696621aff..f96ea2fca 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -45,6 +45,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	int fs, frame_count;
 	AFfilehandle af_fp;
 	int bits;
+	AudioFormat audio_format;
 	mpd_uint16 bitRate;
 	struct stat st;
 
@@ -62,30 +63,30 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, 16);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	dc.audioFormat.bits = (mpd_uint8)bits;
-	dc.audioFormat.sampleRate =
+	audio_format.bits = (mpd_uint8)bits;
+	audio_format.sampleRate =
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
-	dc.audioFormat.channels =
+	audio_format.channels =
 	              (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
-	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
 	dc.totalTime =
-	    ((float)frame_count / (float)dc.audioFormat.sampleRate);
+	    ((float)frame_count / (float)audio_format.sampleRate);
 
 	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5);
 
-	if (dc.audioFormat.bits != 8 && dc.audioFormat.bits != 16) {
+	if (audio_format.bits != 8 && audio_format.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
-		      path, dc.audioFormat.bits);
+		      path, audio_format.bits);
 		afCloseFile(af_fp);
 		return -1;
 	}
 
 	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
 
-	decoder_initialized(decoder);
+	decoder_initialized(decoder, &audio_format);
+
 	{
 		int ret, eof = 0, current = 0;
 		char chunk[CHUNK_SIZE];
@@ -94,7 +95,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 			if (dc.command == DECODE_COMMAND_SEEK) {
 				decoder_clear(decoder);
 				current = dc.seekWhere *
-				    dc.audioFormat.sampleRate;
+				    audio_format.sampleRate;
 				afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
 				dc_command_finished();
 			}
@@ -110,7 +111,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 					     1,
 					     chunk, ret * fs,
 					     (float)current /
-					     (float)dc.audioFormat.
+					     (float)audio_format.
 					     sampleRate, bitRate,
 					     NULL);
 				if (dc.command == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index 4bfc87246..f5d5469f7 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -242,7 +242,7 @@ 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 = (dc.audioFormat.bits / 8);
+	const unsigned int bytes_per_sample = (data->audio_format.bits / 8);
 	const unsigned int bytes_per_channel =
 		bytes_per_sample * frame->header.channels;
 	const unsigned int max_samples = FLAC_CHUNK_SIZE / bytes_per_channel;
@@ -250,7 +250,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
 	float timeChange;
 	FLAC__uint64 newPosition = 0;
 
-	assert(dc.audioFormat.bits > 0);
+	assert(data->audio_format.bits > 0);
 
 	timeChange = ((float)samples) / frame->header.sample_rate;
 	data->time += timeChange;
@@ -415,7 +415,7 @@ static int flac_decode_internal(struct decoder * decoder,
 		}
 	}
 
-	decoder_initialized(decoder);
+	decoder_initialized(decoder, &data.audio_format);
 
 	while (1) {
 		if (!flac_process_single(flacDec))
@@ -424,11 +424,11 @@ static int flac_decode_internal(struct decoder * decoder,
 			break;
 		if (dc.command == DECODE_COMMAND_SEEK) {
 			FLAC__uint64 sampleToSeek = dc.seekWhere *
-			    dc.audioFormat.sampleRate + 0.5;
+			    data.audio_format.sampleRate + 0.5;
 			if (flac_seek_absolute(flacDec, sampleToSeek)) {
 				decoder_clear(decoder);
 				data.time = ((float)sampleToSeek) /
-				    dc.audioFormat.sampleRate;
+				    data.audio_format.sampleRate;
 				data.position = 0;
 			} else
 				dc.seekError = 1;
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 930b041e9..4ab1338bf 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -162,6 +162,7 @@ static void mod_close(mod_Data * data)
 static int mod_decode(struct decoder * decoder, char *path)
 {
 	mod_Data *data;
+	AudioFormat audio_format;
 	float total_time = 0.0;
 	int ret;
 	float secPerByte;
@@ -176,16 +177,16 @@ static int mod_decode(struct decoder * decoder, char *path)
 	}
 
 	dc.totalTime = 0;
-	dc.audioFormat.bits = 16;
-	dc.audioFormat.sampleRate = 44100;
-	dc.audioFormat.channels = 2;
-	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+	audio_format.bits = 16;
+	audio_format.sampleRate = 44100;
+	audio_format.channels = 2;
 
 	secPerByte =
-	    1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) *
-		   (float)dc.audioFormat.sampleRate);
+	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
+		   (float)audio_format.sampleRate);
+
+	decoder_initialized(decoder, &audio_format);
 
-	decoder_initialized(decoder);
 	while (1) {
 		if (dc.command == DECODE_COMMAND_SEEK) {
 			dc.seekError = 1;
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 9959f8601..a7f39a3a4 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -1021,6 +1021,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 	mp3DecodeData data;
 	MpdTag *tag = NULL;
 	ReplayGainInfo *replayGainInfo = NULL;
+	AudioFormat audio_format;
 
 	if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
 	    0) {
@@ -1032,8 +1033,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 		return 0;
 	}
 
-	initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
-	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+	initAudioFormatFromMp3DecodeData(&data, &audio_format);
 
 	dc.totalTime = data.totalTime;
 
@@ -1062,7 +1062,7 @@ static int mp3_decode(struct decoder * decoder, InputStream * inStream)
 		freeMpdTag(tag);
 	}
 
-	decoder_initialized(decoder);
+	decoder_initialized(decoder, &audio_format);
 
 	while (mp3Read(&data, decoder, &replayGainInfo) != DECODE_BREAK) ;
 	/* send last little bit if not DECODE_COMMAND_STOP */
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index f917ec345..d2c0f1b6c 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -88,6 +88,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
+	AudioFormat audio_format;
 	unsigned char *mp4Buffer;
 	unsigned int mp4BufferSize;
 	uint32_t sampleRate;
@@ -139,7 +140,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 #endif
 	faacDecSetConfiguration(decoder, config);
 
-	dc.audioFormat.bits = 16;
+	audio_format.bits = 16;
 
 	mp4Buffer = NULL;
 	mp4BufferSize = 0;
@@ -154,8 +155,8 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		return -1;
 	}
 
-	dc.audioFormat.sampleRate = sampleRate;
-	dc.audioFormat.channels = channels;
+	audio_format.sampleRate = sampleRate;
+	audio_format.channels = channels;
 	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
 	scale = mp4ff_time_scale(mp4fh, track);
 
@@ -245,11 +246,9 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
 			scale = frameInfo.samplerate;
 #endif
-			dc.audioFormat.sampleRate = scale;
-			dc.audioFormat.channels = frameInfo.channels;
-			getOutputAudioFormat(&(dc.audioFormat),
-					     &(ob.audioFormat));
-			decoder_initialized(mpd_decoder);
+			audio_format.sampleRate = scale;
+			audio_format.channels = frameInfo.channels;
+			decoder_initialized(mpd_decoder, &audio_format);
 		}
 
 		if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index 13f2adcee..5a57ea4c8 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -111,6 +111,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	mpc_decoder decoder;
 	mpc_reader reader;
 	mpc_streaminfo info;
+	AudioFormat audio_format;
 
 	MpcCallbackData data;
 
@@ -161,11 +162,9 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 
 	dc.totalTime = mpc_streaminfo_get_length(&info);
 
-	dc.audioFormat.bits = 16;
-	dc.audioFormat.channels = info.channels;
-	dc.audioFormat.sampleRate = info.sample_freq;
-
-	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
+	audio_format.bits = 16;
+	audio_format.channels = info.channels;
+	audio_format.sampleRate = info.sample_freq;
 
 	replayGainInfo = newReplayGainInfo();
 	replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -173,11 +172,11 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	replayGainInfo->trackGain = info.gain_title * 0.01;
 	replayGainInfo->trackPeak = info.peak_title / 32767.0;
 
-	decoder_initialized(mpd_decoder);
+	decoder_initialized(mpd_decoder, &audio_format);
 
 	while (!eof) {
 		if (dc.command == DECODE_COMMAND_SEEK) {
-			samplePos = dc.seekWhere * dc.audioFormat.sampleRate;
+			samplePos = dc.seekWhere * audio_format.sampleRate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
 				decoder_clear(mpd_decoder);
 				s16 = (mpd_sint16 *) chunk;
@@ -210,10 +209,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 
 			if (chunkpos >= MPC_CHUNK_SIZE) {
 				total_time = ((float)samplePos) /
-				    dc.audioFormat.sampleRate;
+				    audio_format.sampleRate;
 
 				bitRate = vbrUpdateBits *
-				    dc.audioFormat.sampleRate / 1152 / 1000;
+				    audio_format.sampleRate / 1152 / 1000;
 
 				decoder_data(mpd_decoder, inStream,
 					     inStream->seekable,
@@ -232,10 +231,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	}
 
 	if (dc.command != DECODE_COMMAND_STOP && chunkpos > 0) {
-		total_time = ((float)samplePos) / dc.audioFormat.sampleRate;
+		total_time = ((float)samplePos) / audio_format.sampleRate;
 
 		bitRate =
-		    vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000;
+		    vbrUpdateBits * audio_format.sampleRate / 1152 / 1000;
 
 		decoder_data(mpd_decoder, NULL, inStream->seekable,
 			     chunk, chunkpos, total_time, bitRate,
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index b5e73e455..5879b7054 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -188,7 +188,7 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(const
 		     c_chan++) {
 			u16 = buf[c_chan][c_samp];
 			uc = (unsigned char *)&u16;
-			for (i = 0; i < (dc.audioFormat.bits / 8); i++) {
+			for (i = 0; i < (data->audio_format.bits / 8); i++) {
 				if (data->chunk_length >= FLAC_CHUNK_SIZE) {
 					if (flacSendChunk(data) < 0) {
 						return
@@ -345,7 +345,7 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		goto fail;
 	}
 
-	decoder_initialized(mpd_decoder);
+	decoder_initialized(mpd_decoder, &data.audio_format);
 
 	while (1) {
 		OggFLAC__seekable_stream_decoder_process_single(decoder);
@@ -353,14 +353,14 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		    OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
 			break;
 		}
-		if (dc.command == DECODE_COMMAND_SEEK) {
-			FLAC__uint64 sampleToSeek = dc.seekWhere *
-			    dc.audioFormat.sampleRate + 0.5;
+		if (dc->command == DECODE_COMMAND_SEEK) {
+			FLAC__uint64 sampleToSeek = dc->seekWhere *
+			    data.audio_format.sampleRate + 0.5;
 			if (OggFLAC__seekable_stream_decoder_seek_absolute
 			    (decoder, sampleToSeek)) {
 				decoder_clear(mpd_decoder);
 				data.time = ((float)sampleToSeek) /
-				    dc.audioFormat.sampleRate;
+				    data.audio_format.sampleRate;
 				data.position = 0;
 			} else
 				dc.seekError = 1;
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index 9e1cf89c0..2fb0a9e52 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -215,6 +215,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 	OggVorbis_File vf;
 	ov_callbacks callbacks;
 	OggCallbackData data;
+	AudioFormat audio_format;
 	int current_section;
 	int prev_section = -1;
 	long ret;
@@ -264,7 +265,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 	dc.totalTime = ov_time_total(&vf, -1);
 	if (dc.totalTime < 0)
 		dc.totalTime = 0;
-	dc.audioFormat.bits = 16;
+	audio_format.bits = 16;
 
 	while (1) {
 		if (dc.command == DECODE_COMMAND_SEEK) {
@@ -281,12 +282,10 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 		if (current_section != prev_section) {
 			/*printf("new song!\n"); */
 			vorbis_info *vi = ov_info(&vf, -1);
-			dc.audioFormat.channels = vi->channels;
-			dc.audioFormat.sampleRate = vi->rate;
+			audio_format.channels = vi->channels;
+			audio_format.sampleRate = vi->rate;
 			if (dc.state == DECODE_STATE_START) {
-				getOutputAudioFormat(&(dc.audioFormat),
-						     &(ob.audioFormat));
-				decoder_initialized(decoder);
+				decoder_initialized(decoder, &audio_format);
 			}
 			comments = ov_comment(&vf, -1)->user_comments;
 			putOggCommentsIntoOutputBuffer(inStream->metaName,
@@ -312,7 +311,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 			decoder_data(decoder, inStream,
 				     inStream->seekable,
 				     chunk, chunkpos,
-				     ov_pcm_tell(&vf) / dc.audioFormat.sampleRate,
+				     ov_pcm_tell(&vf) / audio_format.sampleRate,
 				     bitRate, replayGainInfo);
 			chunkpos = 0;
 			if (dc.command == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index 9d125f053..167278f01 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -128,6 +128,7 @@ static void wavpack_decode(struct decoder * decoder,
                            WavpackContext *wpc, int canseek,
                            ReplayGainInfo *replayGainInfo)
 {
+	AudioFormat audio_format;
 	void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
 	char chunk[CHUNK_SIZE];
 	float file_time;
@@ -136,12 +137,12 @@ static void wavpack_decode(struct decoder * decoder,
 	int position, outsamplesize;
 	int Bps;
 
-	dc.audioFormat.sampleRate = WavpackGetSampleRate(wpc);
-	dc.audioFormat.channels = WavpackGetReducedChannels(wpc);
-	dc.audioFormat.bits = WavpackGetBitsPerSample(wpc);
+	audio_format.sampleRate = WavpackGetSampleRate(wpc);
+	audio_format.channels = WavpackGetReducedChannels(wpc);
+	audio_format.bits = WavpackGetBitsPerSample(wpc);
 
-	if (dc.audioFormat.bits > 16)
-		dc.audioFormat.bits = 16;
+	if (audio_format.bits > 16)
+		audio_format.bits = 16;
 
 	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
 		format_samples = format_samples_float;
@@ -159,16 +160,14 @@ static void wavpack_decode(struct decoder * decoder,
 	outsamplesize = Bps;
 	if (outsamplesize > 2)
 		outsamplesize = 2;
-	outsamplesize *= dc.audioFormat.channels;
+	outsamplesize *= audio_format.channels;
 
-	samplesreq = sizeof(chunk) / (4 * dc.audioFormat.channels);
+	samplesreq = sizeof(chunk) / (4 * audio_format.channels);
 
-	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
-
-	dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
+	dc.totalTime = (float)allsamples / audio_format.sampleRate;
 	dc.seekable = canseek;
 
-	decoder_initialized(decoder);
+	decoder_initialized(decoder, &audio_format);
 
 	position = 0;
 
@@ -180,7 +179,7 @@ static void wavpack_decode(struct decoder * decoder,
 				decoder_clear(decoder);
 
 				where = dc.seekWhere *
-				        dc.audioFormat.sampleRate;
+				        audio_format.sampleRate;
 				if (WavpackSeekSample(wpc, where))
 					position = where;
 				else
@@ -202,10 +201,10 @@ static void wavpack_decode(struct decoder * decoder,
 			              1000 + 0.5);
 			position += samplesgot;
 			file_time = (float)position /
-			            dc.audioFormat.sampleRate;
+			            audio_format.sampleRate;
 
 			format_samples(Bps, chunk,
-			               samplesgot * dc.audioFormat.channels);
+			               samplesgot * audio_format.channels);
 
 			decoder_data(decoder, NULL, 0, chunk,
 				     samplesgot * outsamplesize,