Stop passing our single DecoderControl object everywhere

This at least makes the argument list to a lot of our plugin
functions shorter and removes a good amount of line nois^W^Wcode,
hopefully making things easier to read and follow.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7353 09075e82-0dd4-0310-85a5-a0d7c8717e4f

12 files changed, 222 insertions, 244 deletions

diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 3b351d3a7..a26163303 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -36,15 +36,13 @@
 #include <FLAC/format.h>
 #include <FLAC/metadata.h>
 
-void init_FlacData(FlacData * data, OutputBuffer * cb,
-		   DecoderControl * dc, InputStream * inStream)
+void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream)
 {
 	data->chunk_length = 0;
 	data->time = 0;
 	data->position = 0;
 	data->bitRate = 0;
 	data->cb = cb;
-	data->dc = dc;
 	data->inStream = inStream;
 	data->replayGainInfo = NULL;
 	data->tag = NULL;
@@ -165,16 +163,15 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
 void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 			     FlacData * data)
 {
-	DecoderControl *dc = data->dc;
 	const FLAC__StreamMetadata_StreamInfo *si = &(block->data.stream_info);
 
 	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;
-		dc->totalTime = ((float)si->total_samples) / (si->sample_rate);
-		getOutputAudioFormat(&(dc->audioFormat),
+		dc.audioFormat.bits = (mpd_sint8)si->bits_per_sample;
+		dc.audioFormat.sampleRate = si->sample_rate;
+		dc.audioFormat.channels = (mpd_sint8)si->channels;
+		dc.totalTime = ((float)si->total_samples) / (si->sample_rate);
+		getOutputAudioFormat(&(dc.audioFormat),
 				     &(data->cb->audioFormat));
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
@@ -188,7 +185,7 @@ void flac_error_common_cb(const char *plugin,
 			  const FLAC__StreamDecoderErrorStatus status,
 			  FlacData * data)
 {
-	if (data->dc->stop)
+	if (dc.stop)
 		return;
 
 	switch (status) {
diff --git a/src/inputPlugins/_flac_common.h b/src/inputPlugins/_flac_common.h
index faeccf1ca..37b5fdaae 100644
--- a/src/inputPlugins/_flac_common.h
+++ b/src/inputPlugins/_flac_common.h
@@ -149,15 +149,13 @@ typedef struct {
 	unsigned int bitRate;
 	FLAC__uint64 position;
 	OutputBuffer *cb;
-	DecoderControl *dc;
 	InputStream *inStream;
 	ReplayGainInfo *replayGainInfo;
 	MpdTag *tag;
 } FlacData;
 
 /* initializes a given FlacData struct */
-void init_FlacData(FlacData * data, OutputBuffer * cb,
-		   DecoderControl * dc, InputStream * inStream);
+void init_FlacData(FlacData * data, OutputBuffer * cb, InputStream * inStream);
 void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 			     FlacData * data);
 void flac_error_common_cb(const char *plugin,
@@ -171,7 +169,7 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
 static inline int flacSendChunk(FlacData * data)
 {
 	if (sendDataToOutputBuffer(data->cb, data->inStream,
-	                           data->dc, 1, data->chunk,
+	                           1, data->chunk,
 				   data->chunk_length, data->time,
 				   data->bitRate,
 				   data->replayGainInfo) ==
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index 2962b57c6..aeda10492 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -282,7 +282,7 @@ static int getAacTotalTime(char *file)
 	return file_time;
 }
 
-static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int aac_decode(OutputBuffer * cb, char *path)
 {
 	float file_time;
 	float totalTime;
@@ -339,9 +339,9 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 		return -1;
 	}
 
-	dc->audioFormat.bits = 16;
+	dc.audioFormat.bits = 16;
 
-	dc->totalTime = totalTime;
+	dc.totalTime = totalTime;
 
 	file_time = 0.0;
 
@@ -372,12 +372,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 		sampleRate = frameInfo.samplerate;
 #endif
 
-		if (dc->state != DECODE_STATE_DECODE) {
-			dc->audioFormat.channels = frameInfo.channels;
-			dc->audioFormat.sampleRate = sampleRate;
-			getOutputAudioFormat(&(dc->audioFormat),
+		if (dc.state != DECODE_STATE_DECODE) {
+			dc.audioFormat.channels = frameInfo.channels;
+			dc.audioFormat.sampleRate = sampleRate;
+			getOutputAudioFormat(&(dc.audioFormat),
 					     &(cb->audioFormat));
-			dc->state = DECODE_STATE_DECODE;
+			dc.state = DECODE_STATE_DECODE;
 		}
 
 		advanceAacBuffer(&b, frameInfo.bytesconsumed);
@@ -395,14 +395,14 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 
 		sampleBufferLen = sampleCount * 2;
 
-		sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer,
+		sendDataToOutputBuffer(cb, NULL, 0, sampleBuffer,
 				       sampleBufferLen, file_time,
 				       bitRate, NULL);
-		if (dc->seek) {
-			dc->seekError = 1;
-			dc->seek = 0;
+		if (dc.seek) {
+			dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
-		} else if (dc->stop) {
+		} else if (dc.stop) {
 			eof = 1;
 			break;
 		}
@@ -414,12 +414,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 	if (b.buffer)
 		free(b.buffer);
 
-	if (dc->state != DECODE_STATE_DECODE)
+	if (dc.state != DECODE_STATE_DECODE)
 		return -1;
 
-	if (dc->seek) {
-		dc->seekError = 1;
-		dc->seek = 0;
+	if (dc.seek) {
+		dc.seekError = 1;
+		dc.seek = 0;
 		decoder_wakeup_player();
 	}
 
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 33ea54df9..4510ba46a 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -45,7 +45,7 @@ static int getAudiofileTotalTime(char *file)
 	return total_time;
 }
 
-static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int audiofile_decode(OutputBuffer * cb, char *path)
 {
 	int fs, frame_count;
 	AFfilehandle af_fp;
@@ -67,41 +67,41 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, 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 =
+	dc.audioFormat.bits = (mpd_uint8)bits;
+	dc.audioFormat.sampleRate =
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
-	dc->audioFormat.channels =
+	dc.audioFormat.channels =
 	              (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
-	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
-	dc->totalTime =
-	    ((float)frame_count / (float)dc->audioFormat.sampleRate);
+	dc.totalTime =
+	    ((float)frame_count / (float)dc.audioFormat.sampleRate);
 
-	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5);
+	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.totalTime / 1000.0 + 0.5);
 
-	if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
+	if (dc.audioFormat.bits != 8 && dc.audioFormat.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
-		      path, dc->audioFormat.bits);
+		      path, dc.audioFormat.bits);
 		afCloseFile(af_fp);
 		return -1;
 	}
 
 	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 	{
 		int ret, eof = 0, current = 0;
 		char chunk[CHUNK_SIZE];
 
 		while (!eof) {
-			if (dc->seek) {
+			if (dc.seek) {
 				clearOutputBuffer(cb);
-				current = dc->seekWhere *
-				    dc->audioFormat.sampleRate;
+				current = dc.seekWhere *
+				    dc.audioFormat.sampleRate;
 				afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
-				dc->seek = 0;
+				dc.seek = 0;
 				decoder_wakeup_player();
 			}
 
@@ -114,15 +114,14 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 				current += ret;
 				sendDataToOutputBuffer(cb,
 						       NULL,
-						       dc,
 						       1,
 						       chunk,
 						       ret * fs,
 						       (float)current /
-						       (float)dc->audioFormat.
+						       (float)dc.audioFormat.
 						       sampleRate, bitRate,
 						       NULL);
-				if (dc->stop)
+				if (dc.stop)
 					break;
 			}
 		}
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index b5c18af75..f171ee457 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -41,15 +41,14 @@ static flac_read_status flacRead(const flac_decoder * flacDec,
 
 	while (1) {
 		r = readFromInputStream(data->inStream, (void *)buf, 1, *bytes);
-		if (r == 0 && !inputStreamAtEOF(data->inStream) &&
-		    !data->dc->stop)
+		if (r == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
 			my_usleep(10000);
 		else
 			break;
 	}
 	*bytes = r;
 
-	if (r == 0 && !data->dc->stop) {
+	if (r == 0 && !dc.stop) {
 		if (inputStreamAtEOF(data->inStream))
 			return flac_read_status_eof;
 		else
@@ -248,7 +247,7 @@ 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 = (data->dc->audioFormat.bits / 8);
+	const unsigned int bytes_per_sample = (dc.audioFormat.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;
@@ -256,7 +255,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
 	float timeChange;
 	FLAC__uint64 newPosition = 0;
 
-	assert(data->dc->audioFormat.bits > 0);
+	assert(dc.audioFormat.bits > 0);
 
 	timeChange = ((float)samples) / frame->header.sample_rate;
 	data->time += timeChange;
@@ -292,7 +291,7 @@ static FLAC__StreamDecoderWriteStatus flacWrite(const flac_decoder *dec,
 				FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
 		}
 		data->chunk_length = 0;
-		if (data->dc->seek) {
+		if (dc.seek) {
 			return
 				FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
 		}
@@ -382,7 +381,7 @@ static MpdTag *flacTagDup(char *file)
 	return ret;
 }
 
-static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
+static int flac_decode_internal(OutputBuffer * cb,
                                InputStream * inStream, int is_ogg)
 {
 	flac_decoder *flacDec;
@@ -391,7 +390,7 @@ static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
 
 	if (!(flacDec = flac_new()))
 		return -1;
-	init_FlacData(&data, cb, dc, inStream);
+	init_FlacData(&data, cb, inStream);
 
 #if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
         if(!FLAC__stream_decoder_set_metadata_respond(flacDec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
@@ -421,33 +420,33 @@ static int flac_decode_internal(OutputBuffer * cb, DecoderControl * dc,
 		}
 	}
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 
 	while (1) {
 		if (!flac_process_single(flacDec))
 			break;
 		if (flac_get_state(flacDec) == flac_decoder_eof)
 			break;
-		if (dc->seek) {
-			FLAC__uint64 sampleToSeek = dc->seekWhere *
-			    dc->audioFormat.sampleRate + 0.5;
+		if (dc.seek) {
+			FLAC__uint64 sampleToSeek = dc.seekWhere *
+			    dc.audioFormat.sampleRate + 0.5;
 			if (flac_seek_absolute(flacDec, sampleToSeek)) {
 				clearOutputBuffer(cb);
 				data.time = ((float)sampleToSeek) /
-				    dc->audioFormat.sampleRate;
+				    dc.audioFormat.sampleRate;
 				data.position = 0;
 			} else
-				dc->seekError = 1;
-			dc->seek = 0;
+				dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 	}
-	if (!dc->stop) {
+	if (!dc.stop) {
 		flacPrintErroredState(flac_get_state(flacDec));
 		flac_finish(flacDec);
 	}
 	/* send last little bit */
-	if (data.chunk_length > 0 && !dc->stop) {
+	if (data.chunk_length > 0 && !dc.stop) {
 		flacSendChunk(&data);
 		flushOutputBuffer(data.cb);
 	}
@@ -466,10 +465,9 @@ fail:
 	return 0;
 }
 
-static int flac_decode(OutputBuffer * cb, DecoderControl * dc,
-                       InputStream * inStream)
+static int flac_decode(OutputBuffer * cb, InputStream * inStream)
 {
-	return flac_decode_internal(cb, dc, inStream, 0);
+	return flac_decode_internal(cb, inStream, 0);
 }
 
 #if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
@@ -508,10 +506,9 @@ out:
 	return ret;
 }
 
-static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
-		          InputStream * inStream)
+static int oggflac_decode(OutputBuffer * cb, InputStream * inStream)
 {
-	return flac_decode_internal(cb, dc, inStream, 1);
+	return flac_decode_internal(cb, inStream, 1);
 }
 
 static unsigned int oggflac_try_decode(InputStream * inStream)
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 25cedf04b..728f42d6f 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -163,7 +163,7 @@ static void mod_close(mod_Data * data)
 	free(data);
 }
 
-static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+static int mod_decode(OutputBuffer * cb, char *path)
 {
 	mod_Data *data;
 	float total_time = 0.0;
@@ -179,25 +179,25 @@ static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 		return -1;
 	}
 
-	dc->totalTime = 0;
-	dc->audioFormat.bits = 16;
-	dc->audioFormat.sampleRate = 44100;
-	dc->audioFormat.channels = 2;
-	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+	dc.totalTime = 0;
+	dc.audioFormat.bits = 16;
+	dc.audioFormat.sampleRate = 44100;
+	dc.audioFormat.channels = 2;
+	getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
 
 	secPerByte =
-	    1.0 / ((dc->audioFormat.bits * dc->audioFormat.channels / 8.0) *
-		   (float)dc->audioFormat.sampleRate);
+	    1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) *
+		   (float)dc.audioFormat.sampleRate);
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 	while (1) {
-		if (dc->seek) {
-			dc->seekError = 1;
-			dc->seek = 0;
+		if (dc.seek) {
+			dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 
-		if (dc->stop)
+		if (dc.stop)
 			break;
 
 		if (!Player_Active())
@@ -205,7 +205,7 @@ static int mod_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 
 		ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
 		total_time += ret * secPerByte;
-		sendDataToOutputBuffer(cb, NULL, dc, 0,
+		sendDataToOutputBuffer(cb, NULL, 0,
 				       (char *)data->audio_buffer, ret,
 				       total_time, 0, NULL);
 	}
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 875ec1482..ea33ad5ad 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -674,7 +674,7 @@ static int parse_lame(struct lame *lame, struct mad_bitptr *ptr, int *bitlen)
 	return 1;
 }
 
-static int decodeFirstFrame(mp3DecodeData * data, DecoderControl * dc,
+static int decodeFirstFrame(mp3DecodeData * data,
                             MpdTag ** tag, ReplayGainInfo ** replayGainInfo)
 {
 	struct xing xing;
@@ -689,13 +689,13 @@ static int decodeFirstFrame(mp3DecodeData * data, DecoderControl * dc,
 
 	while (1) {
 		while ((ret = decodeNextFrameHeader(data, tag, replayGainInfo)) == DECODE_CONT &&
-		       (!dc || !dc->stop));
-		if (ret == DECODE_BREAK || (dc && dc->stop)) return -1;
+		       !dc.stop);
+		if (ret == DECODE_BREAK || dc.stop) return -1;
 		if (ret == DECODE_SKIP) continue;
 
 		while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
-		       (!dc || !dc->stop));
-		if (ret == DECODE_BREAK || (dc && dc->stop)) return -1;
+		       !dc.stop);
+		if (ret == DECODE_BREAK || dc.stop) return -1;
 		if (ret == DECODE_OK) break;
 	}
 
@@ -787,7 +787,7 @@ static int getMp3TotalTime(char *file)
 	if (openInputStream(&inStream, file) < 0)
 		return -1;
 	initMp3DecodeData(&data, &inStream);
-	if (decodeFirstFrame(&data, NULL, NULL, NULL) < 0)
+	if (decodeFirstFrame(&data, NULL, NULL) < 0)
 		ret = -1;
 	else
 		ret = data.totalTime + 0.5;
@@ -798,12 +798,12 @@ static int getMp3TotalTime(char *file)
 }
 
 static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
-				  DecoderControl * dc, MpdTag ** tag,
+				  MpdTag ** tag,
 				  ReplayGainInfo ** replayGainInfo)
 {
 	initMp3DecodeData(data, inStream);
 	*tag = NULL;
-	if (decodeFirstFrame(data, dc, tag, replayGainInfo) < 0) {
+	if (decodeFirstFrame(data, tag, replayGainInfo) < 0) {
 		mp3DecodeDataFinalize(data);
 		if (tag && *tag)
 			freeMpdTag(*tag);
@@ -813,7 +813,7 @@ static int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
 	return 0;
 }
 
-static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
+static int mp3Read(mp3DecodeData * data, OutputBuffer * cb,
 		   ReplayGainInfo ** replayGainInfo)
 {
 	int samplesPerFrame;
@@ -852,11 +852,11 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
 		data->muteFrame = 0;
 		break;
 	case MUTEFRAME_SEEK:
-		if (dc->seekWhere <= data->elapsedTime) {
+		if (dc.seekWhere <= data->elapsedTime) {
 			data->outputPtr = data->outputBuffer;
 			clearOutputBuffer(cb);
 			data->muteFrame = 0;
-			dc->seek = 0;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 		break;
@@ -931,7 +931,6 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
 			if (data->outputPtr >= data->outputBufferEnd) {
 				ret = sendDataToOutputBuffer(cb,
 							     data->inStream,
-							     dc,
 							     data->inStream->seekable,
 							     data->outputBuffer,
 							     data->outputPtr - data->outputBuffer,
@@ -952,10 +951,10 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
 
 		data->decodedFirstFrame = 1;
 
-		if (dc->seek && data->inStream->seekable) {
+		if (dc.seek && data->inStream->seekable) {
 			long j = 0;
 			data->muteFrame = MUTEFRAME_SEEK;
-			while (j < data->highestFrame && dc->seekWhere >
+			while (j < data->highestFrame && dc.seekWhere >
 			       ((float)mad_timer_count(data->times[j],
 						       MAD_UNITS_MILLISECONDS))
 			       / 1000) {
@@ -969,14 +968,14 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
 					clearOutputBuffer(cb);
 					data->currentFrame = j;
 				} else
-					dc->seekError = 1;
+					dc.seekError = 1;
 				data->muteFrame = 0;
-				dc->seek = 0;
+				dc.seek = 0;
 				decoder_wakeup_player();
 			}
-		} else if (dc->seek && !data->inStream->seekable) {
-			dc->seek = 0;
-			dc->seekError = 1;
+		} else if (dc.seek && !data->inStream->seekable) {
+			dc.seek = 0;
+			dc.seekError = 1;
 			decoder_wakeup_player();
 		}
 	}
@@ -986,22 +985,22 @@ static int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc,
 		while ((ret =
 			decodeNextFrameHeader(data, NULL,
 					      replayGainInfo)) == DECODE_CONT
-		       && !dc->stop) ;
-		if (ret == DECODE_BREAK || dc->stop || dc->seek)
+		       && !dc.stop) ;
+		if (ret == DECODE_BREAK || dc.stop || dc.seek)
 			break;
 		else if (ret == DECODE_SKIP)
 			skip = 1;
 		if (!data->muteFrame) {
 			while ((ret = decodeNextFrame(data)) == DECODE_CONT &&
-			       !dc->stop && !dc->seek) ;
-			if (ret == DECODE_BREAK || dc->stop || dc->seek)
+			       !dc.stop && !dc.seek) ;
+			if (ret == DECODE_BREAK || dc.stop || dc.seek)
 				break;
 		}
 		if (!skip && ret == DECODE_OK)
 			break;
 	}
 
-	if (dc->stop)
+	if (dc.stop)
 		return DECODE_BREAK;
 
 	return ret;
@@ -1015,16 +1014,15 @@ static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
 	af->channels = MAD_NCHANNELS(&(data->frame).header);
 }
 
-static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
-		      InputStream * inStream)
+static int mp3_decode(OutputBuffer * cb, InputStream * inStream)
 {
 	mp3DecodeData data;
 	MpdTag *tag = NULL;
 	ReplayGainInfo *replayGainInfo = NULL;
 
-	if (openMp3FromInputStream(inStream, &data, dc, &tag, &replayGainInfo) <
+	if (openMp3FromInputStream(inStream, &data, &tag, &replayGainInfo) <
 	    0) {
-		if (!dc->stop) {
+		if (!dc.stop) {
 			ERROR
 			    ("Input does not appear to be a mp3 bit stream.\n");
 			return -1;
@@ -1032,10 +1030,10 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
 		return 0;
 	}
 
-	initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
-	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+	initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
 
-	dc->totalTime = data.totalTime;
+	dc.totalTime = data.totalTime;
 
 	if (inStream->metaTitle) {
 		if (tag)
@@ -1062,12 +1060,12 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
 		freeMpdTag(tag);
 	}
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 
-	while (mp3Read(&data, cb, dc, &replayGainInfo) != DECODE_BREAK) ;
-	/* send last little bit if not dc->stop */
-	if (!dc->stop && data.outputPtr != data.outputBuffer && data.flush) {
-		sendDataToOutputBuffer(cb, NULL, dc,
+	while (mp3Read(&data, cb, &replayGainInfo) != DECODE_BREAK) ;
+	/* send last little bit if not dc.stop */
+	if (!dc.stop && data.outputPtr != data.outputBuffer && data.flush) {
+		sendDataToOutputBuffer(cb, NULL,
 				       data.inStream->seekable,
 				       data.outputBuffer,
 				       data.outputPtr - data.outputBuffer,
@@ -1078,9 +1076,9 @@ static int mp3_decode(OutputBuffer * cb, DecoderControl * dc,
 	if (replayGainInfo)
 		freeReplayGainInfo(replayGainInfo);
 
-	if (dc->seek && data.muteFrame == MUTEFRAME_SEEK) {
+	if (dc.seek && data.muteFrame == MUTEFRAME_SEEK) {
 		clearOutputBuffer(cb);
-		dc->seek = 0;
+		dc.seek = 0;
 		decoder_wakeup_player();
 	}
 
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 1e755b95c..fb8c71020 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -84,8 +84,7 @@ static uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position)
 	return seekInputStream((InputStream *) inStream, position, SEEK_SET);
 }
 
-static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
-                      InputStream * inStream)
+static int mp4_decode(OutputBuffer * cb, InputStream * inStream)
 {
 	mp4ff_t *mp4fh;
 	mp4ff_callback_t *mp4cb;
@@ -146,7 +145,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 #endif
 	faacDecSetConfiguration(decoder, config);
 
-	dc->audioFormat.bits = 16;
+	dc.audioFormat.bits = 16;
 
 	mp4Buffer = NULL;
 	mp4BufferSize = 0;
@@ -161,8 +160,8 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 		return -1;
 	}
 
-	dc->audioFormat.sampleRate = sampleRate;
-	dc->audioFormat.channels = channels;
+	dc.audioFormat.sampleRate = sampleRate;
+	dc.audioFormat.channels = channels;
 	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
 	scale = mp4ff_time_scale(mp4fh, track);
 
@@ -176,7 +175,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 		free(mp4cb);
 		return -1;
 	}
-	dc->totalTime = ((float)file_time) / scale;
+	dc.totalTime = ((float)file_time) / scale;
 
 	numSamples = mp4ff_num_samples(mp4fh, track);
 
@@ -185,13 +184,13 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 	seekTable = xmalloc(sizeof(float) * numSamples);
 
 	for (sampleId = 0; sampleId < numSamples && !eof; sampleId++) {
-		if (dc->seek)
+		if (dc.seek)
 			seeking = 1;
 
 		if (seeking && seekTableEnd > 1 &&
-		    seekTable[seekTableEnd] >= dc->seekWhere) {
+		    seekTable[seekTableEnd] >= dc.seekWhere) {
 			int i = 2;
-			while (seekTable[i] < dc->seekWhere)
+			while (seekTable[i] < dc.seekWhere)
 				i++;
 			sampleId = i - 1;
 			file_time = seekTable[sampleId];
@@ -213,14 +212,14 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 			dur -= offset;
 		file_time += ((float)dur) / scale;
 
-		if (seeking && file_time > dc->seekWhere)
+		if (seeking && file_time > dc.seekWhere)
 			seekPositionFound = 1;
 
 		if (seeking && seekPositionFound) {
 			seekPositionFound = 0;
 			clearOutputBuffer(cb);
 			seeking = 0;
-			dc->seek = 0;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 
@@ -248,16 +247,16 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 			break;
 		}
 
-		if (dc->state != DECODE_STATE_DECODE) {
+		if (dc.state != DECODE_STATE_DECODE) {
 			channels = frameInfo.channels;
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
 			scale = frameInfo.samplerate;
 #endif
-			dc->audioFormat.sampleRate = scale;
-			dc->audioFormat.channels = frameInfo.channels;
-			getOutputAudioFormat(&(dc->audioFormat),
+			dc.audioFormat.sampleRate = scale;
+			dc.audioFormat.channels = frameInfo.channels;
+			getOutputAudioFormat(&(dc.audioFormat),
 					     &(cb->audioFormat));
-			dc->state = DECODE_STATE_DECODE;
+			dc.state = DECODE_STATE_DECODE;
 		}
 
 		if (channels * (unsigned long)(dur + offset) > frameInfo.samples) {
@@ -278,10 +277,10 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 
 		sampleBuffer += offset * channels * 2;
 
-		sendDataToOutputBuffer(cb, inStream, dc, 1, sampleBuffer,
+		sendDataToOutputBuffer(cb, inStream, 1, sampleBuffer,
 				       sampleBufferLen, file_time,
 				       bitRate, NULL);
-		if (dc->stop) {
+		if (dc.stop) {
 			eof = 1;
 			break;
 		}
@@ -292,12 +291,12 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 	mp4ff_close(mp4fh);
 	free(mp4cb);
 
-	if (dc->state != DECODE_STATE_DECODE)
+	if (dc.state != DECODE_STATE_DECODE)
 		return -1;
 
-	if (dc->seek && seeking) {
+	if (dc.seek && seeking) {
 		clearOutputBuffer(cb);
-		dc->seek = 0;
+		dc.seek = 0;
 		decoder_wakeup_player();
 	}
 	flushOutputBuffer(cb);
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index 9b6e862ff..867965688 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -33,7 +33,6 @@
 
 typedef struct _MpcCallbackData {
 	InputStream *inStream;
-	DecoderControl *dc;
 } MpcCallbackData;
 
 static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
@@ -43,10 +42,9 @@ static mpc_int32_t mpc_read_cb(void *vdata, void *ptr, mpc_int32_t size)
 
 	while (1) {
 		ret = readFromInputStream(data->inStream, ptr, 1, size);
-		if (ret == 0 && !inputStreamAtEOF(data->inStream) &&
-		    (data->dc && !data->dc->stop)) {
+		if (ret == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
 			my_usleep(10000);
-		} else
+		else
 			break;
 	}
 
@@ -113,8 +111,7 @@ static inline mpd_sint16 convertSample(MPC_SAMPLE_FORMAT sample)
 	return val;
 }
 
-static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
-		      InputStream * inStream)
+static int mpc_decode(OutputBuffer * cb, InputStream * inStream)
 {
 	mpc_decoder decoder;
 	mpc_reader reader;
@@ -139,7 +136,6 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 	ReplayGainInfo *replayGainInfo = NULL;
 
 	data.inStream = inStream;
-	data.dc = dc;
 
 	reader.read = mpc_read_cb;
 	reader.seek = mpc_seek_cb;
@@ -151,7 +147,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 	mpc_streaminfo_init(&info);
 
 	if ((ret = mpc_streaminfo_read(&info, &reader)) != ERROR_CODE_OK) {
-		if (!dc->stop) {
+		if (!dc.stop) {
 			ERROR("Not a valid musepack stream\n");
 			return -1;
 		}
@@ -161,20 +157,20 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 	mpc_decoder_setup(&decoder, &reader);
 
 	if (!mpc_decoder_initialize(&decoder, &info)) {
-		if (!dc->stop) {
+		if (!dc.stop) {
 			ERROR("Not a valid musepack stream\n");
 			return -1;
 		}
 		return 0;
 	}
 
-	dc->totalTime = mpc_streaminfo_get_length(&info);
+	dc.totalTime = mpc_streaminfo_get_length(&info);
 
-	dc->audioFormat.bits = 16;
-	dc->audioFormat.channels = info.channels;
-	dc->audioFormat.sampleRate = info.sample_freq;
+	dc.audioFormat.bits = 16;
+	dc.audioFormat.channels = info.channels;
+	dc.audioFormat.sampleRate = info.sample_freq;
 
-	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
 
 	replayGainInfo = newReplayGainInfo();
 	replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -182,18 +178,18 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 	replayGainInfo->trackGain = info.gain_title * 0.01;
 	replayGainInfo->trackPeak = info.peak_title / 32767.0;
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 
 	while (!eof) {
-		if (dc->seek) {
-			samplePos = dc->seekWhere * dc->audioFormat.sampleRate;
+		if (dc.seek) {
+			samplePos = dc.seekWhere * dc.audioFormat.sampleRate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
 				clearOutputBuffer(cb);
 				s16 = (mpd_sint16 *) chunk;
 				chunkpos = 0;
 			} else
-				dc->seekError = 1;
-			dc->seek = 0;
+				dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 
@@ -202,7 +198,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 		ret = mpc_decoder_decode(&decoder, sample_buffer,
 					 &vbrUpdateAcc, &vbrUpdateBits);
 
-		if (ret <= 0 || dc->stop) {
+		if (ret <= 0 || dc.stop) {
 			eof = 1;
 			break;
 		}
@@ -220,12 +216,12 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 
 			if (chunkpos >= MPC_CHUNK_SIZE) {
 				total_time = ((float)samplePos) /
-				    dc->audioFormat.sampleRate;
+				    dc.audioFormat.sampleRate;
 
 				bitRate = vbrUpdateBits *
-				    dc->audioFormat.sampleRate / 1152 / 1000;
+				    dc.audioFormat.sampleRate / 1152 / 1000;
 
-				sendDataToOutputBuffer(cb, inStream, dc,
+				sendDataToOutputBuffer(cb, inStream,
 						       inStream->seekable,
 						       chunk, chunkpos,
 						       total_time,
@@ -233,7 +229,7 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 
 				chunkpos = 0;
 				s16 = (mpd_sint16 *) chunk;
-				if (dc->stop) {
+				if (dc.stop) {
 					eof = 1;
 					break;
 				}
@@ -241,13 +237,13 @@ static int mpc_decode(OutputBuffer * cb, DecoderControl * dc,
 		}
 	}
 
-	if (!dc->stop && chunkpos > 0) {
-		total_time = ((float)samplePos) / dc->audioFormat.sampleRate;
+	if (!dc.stop && chunkpos > 0) {
+		total_time = ((float)samplePos) / dc.audioFormat.sampleRate;
 
 		bitRate =
-		    vbrUpdateBits * dc->audioFormat.sampleRate / 1152 / 1000;
+		    vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000;
 
-		sendDataToOutputBuffer(cb, NULL, dc, inStream->seekable,
+		sendDataToOutputBuffer(cb, NULL, inStream->seekable,
 				       chunk, chunkpos, total_time, bitRate,
 				       replayGainInfo);
 	}
@@ -269,7 +265,6 @@ static float mpcGetTime(char *file)
 	MpcCallbackData data;
 
 	data.inStream = &inStream;
-	data.dc = NULL;
 
 	reader.read = mpc_read_cb;
 	reader.seek = mpc_seek_cb;
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index 6638362b6..070404e26 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -56,14 +56,14 @@ static OggFLAC__SeekableStreamDecoderReadStatus of_read_cb(const
 	while (1) {
 		r = readFromInputStream(data->inStream, (void *)buf, 1, *bytes);
 		if (r == 0 && !inputStreamAtEOF(data->inStream) &&
-		    !data->dc->stop)
+		    !dc.stop)
 			my_usleep(10000);
 		else
 			break;
 	}
 	*bytes = r;
 
-	if (r == 0 && !inputStreamAtEOF(data->inStream) && !data->dc->stop)
+	if (r == 0 && !inputStreamAtEOF(data->inStream) && !dc.stop)
 		return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_ERROR;
 
 	return OggFLAC__SEEKABLE_STREAM_DECODER_READ_STATUS_OK;
@@ -193,14 +193,14 @@ static FLAC__StreamDecoderWriteStatus oggflacWrite(const
 		     c_chan++) {
 			u16 = buf[c_chan][c_samp];
 			uc = (unsigned char *)&u16;
-			for (i = 0; i < (data->dc->audioFormat.bits / 8); i++) {
+			for (i = 0; i < (dc.audioFormat.bits / 8); i++) {
 				if (data->chunk_length >= FLAC_CHUNK_SIZE) {
 					if (flacSendChunk(data) < 0) {
 						return
 						    FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
 					}
 					data->chunk_length = 0;
-					if (data->dc->seek) {
+					if (dc.seek) {
 						return
 						    FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
 					}
@@ -336,21 +336,20 @@ static unsigned int oggflac_try_decode(InputStream * inStream)
 	return (ogg_stream_type_detect(inStream) == FLAC) ? 1 : 0;
 }
 
-static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
-			  InputStream * inStream)
+static int oggflac_decode(OutputBuffer * cb, InputStream * inStream)
 {
 	OggFLAC__SeekableStreamDecoder *decoder = NULL;
 	FlacData data;
 	int ret = 0;
 
-	init_FlacData(&data, cb, dc, inStream);
+	init_FlacData(&data, cb, inStream);
 
 	if (!(decoder = full_decoder_init_and_read_metadata(&data, 0))) {
 		ret = -1;
 		goto fail;
 	}
 
-	dc->state = DECODE_STATE_DECODE;
+	dc.state = DECODE_STATE_DECODE;
 
 	while (1) {
 		OggFLAC__seekable_stream_decoder_process_single(decoder);
@@ -358,29 +357,29 @@ static int oggflac_decode(OutputBuffer * cb, DecoderControl * dc,
 		    OggFLAC__SEEKABLE_STREAM_DECODER_OK) {
 			break;
 		}
-		if (dc->seek) {
-			FLAC__uint64 sampleToSeek = dc->seekWhere *
-			    dc->audioFormat.sampleRate + 0.5;
+		if (dc.seek) {
+			FLAC__uint64 sampleToSeek = dc.seekWhere *
+			    dc.audioFormat.sampleRate + 0.5;
 			if (OggFLAC__seekable_stream_decoder_seek_absolute
 			    (decoder, sampleToSeek)) {
 				clearOutputBuffer(cb);
 				data.time = ((float)sampleToSeek) /
-				    dc->audioFormat.sampleRate;
+				    dc.audioFormat.sampleRate;
 				data.position = 0;
 			} else
-				dc->seekError = 1;
-			dc->seek = 0;
+				dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 	}
 
-	if (!dc->stop) {
+	if (!dc.stop) {
 		oggflacPrintErroredState
 		    (OggFLAC__seekable_stream_decoder_get_state(decoder));
 		OggFLAC__seekable_stream_decoder_finish(decoder);
 	}
 	/* send last little bit */
-	if (data.chunk_length > 0 && !dc->stop) {
+	if (data.chunk_length > 0 && !dc.stop) {
 		flacSendChunk(&data);
 		flushOutputBuffer(data.cb);
 	}
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index c79e7893c..afcef3e08 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -56,7 +56,6 @@
 
 typedef struct _OggCallbackData {
 	InputStream *inStream;
-	DecoderControl *dc;
 } OggCallbackData;
 
 static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
@@ -67,7 +66,7 @@ static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
 	while (1) {
 		ret = readFromInputStream(data->inStream, ptr, size, nmemb);
 		if (ret == 0 && !inputStreamAtEOF(data->inStream) &&
-		    !data->dc->stop) {
+		    !dc.stop) {
 			my_usleep(10000);
 		} else
 			break;
@@ -81,7 +80,7 @@ static size_t ogg_read_cb(void *ptr, size_t size, size_t nmemb, void *vdata)
 static int ogg_seek_cb(void *vdata, ogg_int64_t offset, int whence)
 {
 	const OggCallbackData *data = (const OggCallbackData *) vdata;
-	if(data->dc->stop)
+	if (dc.stop)
 		return -1;
 	return seekInputStream(data->inStream, offset, whence);
 }
@@ -217,8 +216,7 @@ static void putOggCommentsIntoOutputBuffer(OutputBuffer * cb, char *streamName,
 }
 
 /* public */
-static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
-			    InputStream * inStream)
+static int oggvorbis_decode(OutputBuffer * cb, InputStream * inStream)
 {
 	OggVorbis_File vf;
 	ov_callbacks callbacks;
@@ -236,14 +234,13 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
 	const char *errorStr;
 
 	data.inStream = inStream;
-	data.dc = dc;
 
 	callbacks.read_func = ogg_read_cb;
 	callbacks.seek_func = ogg_seek_cb;
 	callbacks.close_func = ogg_close_cb;
 	callbacks.tell_func = ogg_tell_cb;
 	if ((ret = ov_open_callbacks(&data, &vf, NULL, 0, callbacks)) < 0) {
-		if (!dc->stop) {
+		if (!dc.stop) {
 			switch (ret) {
 			case OV_EREAD:
 				errorStr = "read error";
@@ -270,19 +267,19 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
 		}
 		return 0;
 	}
-	dc->totalTime = ov_time_total(&vf, -1);
-	if (dc->totalTime < 0)
-		dc->totalTime = 0;
-	dc->audioFormat.bits = 16;
+	dc.totalTime = ov_time_total(&vf, -1);
+	if (dc.totalTime < 0)
+		dc.totalTime = 0;
+	dc.audioFormat.bits = 16;
 
 	while (1) {
-		if (dc->seek) {
-			if (0 == ov_time_seek_page(&vf, dc->seekWhere)) {
+		if (dc.seek) {
+			if (0 == ov_time_seek_page(&vf, dc.seekWhere)) {
 				clearOutputBuffer(cb);
 				chunkpos = 0;
 			} else
-				dc->seekError = 1;
-			dc->seek = 0;
+				dc.seekError = 1;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 		ret = ov_read(&vf, chunk + chunkpos,
@@ -291,12 +288,12 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
 		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;
-			if (dc->state == DECODE_STATE_START) {
-				getOutputAudioFormat(&(dc->audioFormat),
+			dc.audioFormat.channels = vi->channels;
+			dc.audioFormat.sampleRate = vi->rate;
+			if (dc.state == DECODE_STATE_START) {
+				getOutputAudioFormat(&(dc.audioFormat),
 						     &(cb->audioFormat));
-				dc->state = DECODE_STATE_DECODE;
+				dc.state = DECODE_STATE_DECODE;
 			}
 			comments = ov_comment(&vf, -1)->user_comments;
 			putOggCommentsIntoOutputBuffer(cb, inStream->metaName,
@@ -319,20 +316,20 @@ static int oggvorbis_decode(OutputBuffer * cb, DecoderControl * dc,
 			if ((test = ov_bitrate_instant(&vf)) > 0) {
 				bitRate = test / 1000;
 			}
-			sendDataToOutputBuffer(cb, inStream, dc,
+			sendDataToOutputBuffer(cb, inStream,
 					       inStream->seekable,
 					       chunk, chunkpos,
 					       ov_pcm_tell(&vf) /
-					       dc->audioFormat.sampleRate,
+					       dc.audioFormat.sampleRate,
 					       bitRate, replayGainInfo);
 			chunkpos = 0;
-			if (dc->stop)
+			if (dc.stop)
 				break;
 		}
 	}
 
-	if (!dc->stop && chunkpos > 0) {
-		sendDataToOutputBuffer(cb, NULL, dc, inStream->seekable,
+	if (!dc.stop && chunkpos > 0) {
+		sendDataToOutputBuffer(cb, NULL, inStream->seekable,
 				       chunk, chunkpos,
 				       ov_time_tell(&vf), bitRate,
 				       replayGainInfo);
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index c46cdeb75..bae5f6acb 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -128,7 +128,7 @@ static void format_samples_float(int Bps, void *buffer, uint32_t samcnt)
  * This does the main decoding thing.
  * Requires an already opened WavpackContext.
  */
-static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
+static void wavpack_decode(OutputBuffer *cb,
                            WavpackContext *wpc, int canseek,
                            ReplayGainInfo *replayGainInfo)
 {
@@ -140,12 +140,12 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
 	int position, outsamplesize;
 	int Bps;
 
-	dc->audioFormat.sampleRate = WavpackGetSampleRate(wpc);
-	dc->audioFormat.channels = WavpackGetReducedChannels(wpc);
-	dc->audioFormat.bits = WavpackGetBitsPerSample(wpc);
+	dc.audioFormat.sampleRate = WavpackGetSampleRate(wpc);
+	dc.audioFormat.channels = WavpackGetReducedChannels(wpc);
+	dc.audioFormat.bits = WavpackGetBitsPerSample(wpc);
 
-	if (dc->audioFormat.bits > 16)
-		dc->audioFormat.bits = 16;
+	if (dc.audioFormat.bits > 16)
+		dc.audioFormat.bits = 16;
 
 	if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT)
 		format_samples = format_samples_float;
@@ -163,40 +163,40 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
 	outsamplesize = Bps;
 	if (outsamplesize > 2)
 		outsamplesize = 2;
-	outsamplesize *= dc->audioFormat.channels;
+	outsamplesize *= dc.audioFormat.channels;
 
-	samplesreq = sizeof(chunk) / (4 * dc->audioFormat.channels);
+	samplesreq = sizeof(chunk) / (4 * dc.audioFormat.channels);
 
-	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(cb->audioFormat));
 
-	dc->totalTime = (float)allsamples / dc->audioFormat.sampleRate;
-	dc->state = DECODE_STATE_DECODE;
-	dc->seekable = canseek;
+	dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
+	dc.state = DECODE_STATE_DECODE;
+	dc.seekable = canseek;
 
 	position = 0;
 
 	do {
-		if (dc->seek) {
+		if (dc.seek) {
 			if (canseek) {
 				int where;
 
 				clearOutputBuffer(cb);
 
-				where = dc->seekWhere *
-				        dc->audioFormat.sampleRate;
+				where = dc.seekWhere *
+				        dc.audioFormat.sampleRate;
 				if (WavpackSeekSample(wpc, where))
 					position = where;
 				else
-					dc->seekError = 1;
+					dc.seekError = 1;
 			} else {
-				dc->seekError = 1;
+				dc.seekError = 1;
 			}
 
-			dc->seek = 0;
+			dc.seek = 0;
 			decoder_wakeup_player();
 		}
 
-		if (dc->stop)
+		if (dc.stop)
 			break;
 
 		samplesgot = WavpackUnpackSamples(wpc,
@@ -206,12 +206,12 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
 			              1000 + 0.5);
 			position += samplesgot;
 			file_time = (float)position /
-			            dc->audioFormat.sampleRate;
+			            dc.audioFormat.sampleRate;
 
 			format_samples(Bps, chunk,
-			               samplesgot * dc->audioFormat.channels);
+			               samplesgot * dc.audioFormat.channels);
 
-			sendDataToOutputBuffer(cb, NULL, dc, 0, chunk,
+			sendDataToOutputBuffer(cb, NULL, 0, chunk,
 			                       samplesgot * outsamplesize,
 			                       file_time, bitrate,
 					       replayGainInfo);
@@ -442,8 +442,7 @@ static unsigned int wavpack_trydecode(InputStream *is)
 /*
  * Decodes a stream.
  */
-static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
-                                InputStream *is)
+static int wavpack_streamdecode(OutputBuffer *cb, InputStream *is)
 {
 	char error[ERRORLEN];
 	WavpackContext *wpc;
@@ -462,7 +461,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
 		err = 1;
 
 		/*
-		 * As we use dc->utf8url, this function will be bad for
+		 * As we use dc.utf8url, this function will be bad for
 		 * single files. utf8url is not absolute file path :/
 		 */
 		utf8url = get_song_url(tmp, pc.current_song);
@@ -507,7 +506,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
 				break;
 			}
 
-			if (dc->stop) {
+			if (dc.stop) {
 				break;
 			}
 
@@ -542,7 +541,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
 		return -1;
 	}
 
-	wavpack_decode(cb, dc, wpc, canseek, NULL);
+	wavpack_decode(cb, wpc, canseek, NULL);
 
 	WavpackCloseFile(wpc);
 	if (wvc_url != NULL) {
@@ -557,7 +556,7 @@ static int wavpack_streamdecode(OutputBuffer *cb, DecoderControl *dc,
 /*
  * Decodes a file.
  */
-static int wavpack_filedecode(OutputBuffer *cb, DecoderControl *dc, char *fname)
+static int wavpack_filedecode(OutputBuffer *cb, char *fname)
 {
 	char error[ERRORLEN];
 	WavpackContext *wpc;
@@ -573,7 +572,7 @@ static int wavpack_filedecode(OutputBuffer *cb, DecoderControl *dc, char *fname)
 
 	replayGainInfo = wavpack_replaygain(wpc);
 
-	wavpack_decode(cb, dc, wpc, 1, replayGainInfo);
+	wavpack_decode(cb, wpc, 1, replayGainInfo);
 
 	if (replayGainInfo)
 		freeReplayGainInfo(replayGainInfo);