Make the OutputBuffer API more consistent

We had functions names varied between
outputBufferFoo, fooOutputBuffer, and output_buffer_foo

That was too confusing for my little brain to handle.
And the global variable was somehow named 'cb' instead of
the more obvious 'ob'...

git-svn-id: https://svn.musicpd.org/mpd/trunk@7355 09075e82-0dd4-0310-85a5-a0d7c8717e4f

17 files changed, 146 insertions, 146 deletions

diff --git a/src/decode.c b/src/decode.c
index 1e8700019..a5110f752 100644
--- a/src/decode.c
+++ b/src/decode.c
@@ -85,7 +85,7 @@ static unsigned calculateCrossFadeChunks(AudioFormat * af, float totalTime)
 	chunks = (af->sampleRate * af->bits * af->channels / 8.0 / CHUNK_SIZE);
 	chunks = (chunks * pc.crossFade + 0.5);
 
-	buffered_chunks = cb.size;
+	buffered_chunks = ob.size;
 	assert(buffered_chunks >= buffered_before_play);
 	if (chunks > (buffered_chunks - buffered_before_play))
 		chunks = buffered_chunks - buffered_before_play;
@@ -124,7 +124,7 @@ static int decodeSeek(int *decodeWaitedOn, int *next)
 	    dc.current_song != pc.current_song) {
 		stopDecode();
 		*next = -1;
-		clearOutputBuffer();
+		ob_clear();
 		dc.error = DECODE_ERROR_NOERROR;
 		dc.start = 1;
 		waitOnDecode(decodeWaitedOn);
@@ -344,7 +344,7 @@ void decoderInit(void)
 		FATAL("Failed to spawn decoder task: %s\n", strerror(errno));
 }
 
-static void crossFade(OutputBufferChunk * a, OutputBufferChunk * b,
+static void crossFade(ob_chunk * a, ob_chunk * b,
 		      AudioFormat * format,
 		      unsigned int fadePosition, unsigned int crossFadeChunks)
 {
@@ -361,7 +361,7 @@ static void crossFade(OutputBufferChunk * a, OutputBufferChunk * b,
 		a->chunkSize = b->chunkSize;
 }
 
-static int playChunk(OutputBufferChunk * chunk,
+static int playChunk(ob_chunk * chunk,
 		     AudioFormat * format, double sizeToTime)
 {
 	pc.elapsedTime = chunk->times;
@@ -413,7 +413,7 @@ static void decodeParent(void)
 		}
 
 		if (buffering) {
-			if (availableOutputBuffer() < bbp) {
+			if (ob_available() < bbp) {
 				/* not enough decoded buffer space yet */
 				player_sleep();
 				continue;
@@ -427,7 +427,7 @@ static void decodeParent(void)
 			   dc.error==DECODE_ERROR_NOERROR) {
 				/* the decoder is ready and ok */
 				decodeWaitedOn = 0;
-				if(openAudioDevice(&(cb.audioFormat))<0) {
+				if(openAudioDevice(&(ob.audioFormat))<0) {
 					char tmp[MPD_PATH_MAX];
 					pc.errored_song = pc.current_song;
 					pc.error = PLAYER_ERROR_AUDIO;
@@ -446,7 +446,7 @@ static void decodeParent(void)
 				pc.sampleRate = dc.audioFormat.sampleRate;
 				pc.bits = dc.audioFormat.bits;
 				pc.channels = dc.audioFormat.channels;
-				sizeToTime = audioFormatSizeToTime(&cb.audioFormat);
+				sizeToTime = audioFormatSizeToTime(&ob.audioFormat);
 			}
 			else if(dc.state!=DECODE_STATE_START) {
 				/* the decoder failed */
@@ -467,7 +467,7 @@ static void decodeParent(void)
 		    pc.queueLockState == PLAYER_QUEUE_UNLOCKED) {
 			/* the decoder has finished the current song;
 			   make it decode the next song */
-			next = cb.end;
+			next = ob.end;
 			dc.start = 1;
 			pc.queueState = PLAYER_QUEUE_DECODE;
 			wakeup_main_task();
@@ -479,7 +479,7 @@ static void decodeParent(void)
 			   calculate how many chunks will be required
 			   for it */
 			crossFadeChunks =
-				calculateCrossFadeChunks(&(cb.audioFormat),
+				calculateCrossFadeChunks(&(ob.audioFormat),
 							 dc.totalTime);
 			if (crossFadeChunks > 0) {
 				doCrossFade = 1;
@@ -492,12 +492,12 @@ static void decodeParent(void)
 
 		if (do_pause)
 			player_sleep();
-		else if (!outputBufferEmpty() && (int)cb.begin != next) {
-			OutputBufferChunk *beginChunk =
-				outputBufferGetChunk(cb.begin);
+		else if (!ob_is_empty() && (int)ob.begin != next) {
+			ob_chunk *beginChunk =
+				ob_get_chunk(ob.begin);
 			unsigned int fadePosition;
 			if (doCrossFade == 1 && next >= 0 &&
-			    (fadePosition = outputBufferRelative(next))
+			    (fadePosition = ob_relative(next))
 			    <= crossFadeChunks) {
 				/* perform cross fade */
 				if (nextChunk < 0) {
@@ -508,11 +508,11 @@ static void decodeParent(void)
 					   chunks in the old song */
 					crossFadeChunks = fadePosition;
 				}
-				nextChunk = outputBufferAbsolute(crossFadeChunks);
+				nextChunk = ob_absolute(crossFadeChunks);
 				if (nextChunk >= 0) {
 					crossFade(beginChunk,
-						  outputBufferGetChunk(nextChunk),
-						  &(cb.audioFormat),
+						  ob_get_chunk(nextChunk),
+						  &(ob.audioFormat),
 						  fadePosition,
 						  crossFadeChunks);
 				} else {
@@ -533,19 +533,19 @@ static void decodeParent(void)
 			}
 
 			/* play the current chunk */
-			if (playChunk(beginChunk, &(cb.audioFormat),
+			if (playChunk(beginChunk, &(ob.audioFormat),
 				      sizeToTime) < 0)
 				break;
-			outputBufferShift();
+			ob_shift();
 			player_wakeup_decoder_nb();
-		} else if (!outputBufferEmpty() && (int)cb.begin == next) {
+		} else if (!ob_is_empty() && (int)ob.begin == next) {
 			/* at the beginning of a new song */
 
 			if (doCrossFade == 1 && nextChunk >= 0) {
 				/* the cross-fade is finished; skip
 				   the section which was cross-faded
 				   (and thus already played) */
-				output_buffer_skip(crossFadeChunks);
+				ob_skip(crossFadeChunks);
 			}
 
 			doCrossFade = 0;
@@ -584,7 +584,7 @@ static void decodeParent(void)
  */
 void decode(void)
 {
-	clearOutputBuffer();
+	ob_clear();
 
 	dc.error = DECODE_ERROR_NOERROR;
 	dc.seek = 0;
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 80b1210d1..cf23a5e8c 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -170,7 +170,7 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 		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), &(cb.audioFormat));
+		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 18e51d587..10c2f3d38 100644
--- a/src/inputPlugins/_flac_common.h
+++ b/src/inputPlugins/_flac_common.h
@@ -167,7 +167,7 @@ MpdTag *copyVorbisCommentBlockToMpdTag(const FLAC__StreamMetadata * block,
 /* keep this inlined, this is just macro but prettier :) */
 static inline int flacSendChunk(FlacData * data)
 {
-	if (sendDataToOutputBuffer(data->inStream,
+	if (ob_send(data->inStream,
 	                           1, data->chunk,
 				   data->chunk_length, data->time,
 				   data->bitRate,
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index ebf402be1..6e53c6420 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -376,7 +376,7 @@ static int aac_decode(char *path)
 			dc.audioFormat.channels = frameInfo.channels;
 			dc.audioFormat.sampleRate = sampleRate;
 			getOutputAudioFormat(&(dc.audioFormat),
-					     &(cb.audioFormat));
+					     &(ob.audioFormat));
 			dc.state = DECODE_STATE_DECODE;
 		}
 
@@ -395,7 +395,7 @@ static int aac_decode(char *path)
 
 		sampleBufferLen = sampleCount * 2;
 
-		sendDataToOutputBuffer(NULL, 0, sampleBuffer,
+		ob_send(NULL, 0, sampleBuffer,
 				       sampleBufferLen, file_time,
 				       bitRate, NULL);
 		if (dc.seek) {
@@ -408,7 +408,7 @@ static int aac_decode(char *path)
 		}
 	}
 
-	flushOutputBuffer();
+	ob_flush();
 
 	faacDecClose(decoder);
 	if (b.buffer)
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index d661278b1..558731dd3 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -72,7 +72,7 @@ static int audiofile_decode(char *path)
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
 	dc.audioFormat.channels =
 	              (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
-	getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
@@ -97,7 +97,7 @@ static int audiofile_decode(char *path)
 
 		while (!eof) {
 			if (dc.seek) {
-				clearOutputBuffer();
+				ob_clear();
 				current = dc.seekWhere *
 				    dc.audioFormat.sampleRate;
 				afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
@@ -112,7 +112,7 @@ static int audiofile_decode(char *path)
 				eof = 1;
 			else {
 				current += ret;
-				sendDataToOutputBuffer(NULL,
+				ob_send(NULL,
 						       1,
 						       chunk,
 						       ret * fs,
@@ -125,7 +125,7 @@ static int audiofile_decode(char *path)
 			}
 		}
 
-		flushOutputBuffer();
+		ob_flush();
 	}
 	afCloseFile(af_fp);
 
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index 70b5c7a80..38131bac9 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -430,7 +430,7 @@ static int flac_decode_internal(InputStream * inStream, int is_ogg)
 			FLAC__uint64 sampleToSeek = dc.seekWhere *
 			    dc.audioFormat.sampleRate + 0.5;
 			if (flac_seek_absolute(flacDec, sampleToSeek)) {
-				clearOutputBuffer();
+				ob_clear();
 				data.time = ((float)sampleToSeek) /
 				    dc.audioFormat.sampleRate;
 				data.position = 0;
@@ -447,7 +447,7 @@ static int flac_decode_internal(InputStream * inStream, int is_ogg)
 	/* send last little bit */
 	if (data.chunk_length > 0 && !dc.stop) {
 		flacSendChunk(&data);
-		flushOutputBuffer();
+		ob_flush();
 	}
 
 fail:
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 31ffa9a3d..4b79a3672 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -183,7 +183,7 @@ static int mod_decode(char *path)
 	dc.audioFormat.bits = 16;
 	dc.audioFormat.sampleRate = 44100;
 	dc.audioFormat.channels = 2;
-	getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	secPerByte =
 	    1.0 / ((dc.audioFormat.bits * dc.audioFormat.channels / 8.0) *
@@ -205,12 +205,12 @@ static int mod_decode(char *path)
 
 		ret = VC_WriteBytes(data->audio_buffer, MIKMOD_FRAME_SIZE);
 		total_time += ret * secPerByte;
-		sendDataToOutputBuffer(NULL, 0,
+		ob_send(NULL, 0,
 				       (char *)data->audio_buffer, ret,
 				       total_time, 0, NULL);
 	}
 
-	flushOutputBuffer();
+	ob_flush();
 
 	mod_close(data);
 
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index ee26385d9..dcfc25cdc 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -853,7 +853,7 @@ static int mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
 	case MUTEFRAME_SEEK:
 		if (dc.seekWhere <= data->elapsedTime) {
 			data->outputPtr = data->outputBuffer;
-			clearOutputBuffer();
+			ob_clear();
 			data->muteFrame = 0;
 			dc.seek = 0;
 			decoder_wakeup_player();
@@ -928,7 +928,7 @@ static int mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
 			}
 
 			if (data->outputPtr >= data->outputBufferEnd) {
-				ret = sendDataToOutputBuffer(data->inStream,
+				ret = ob_send(data->inStream,
 							     data->inStream->seekable,
 							     data->outputBuffer,
 							     data->outputPtr - data->outputBuffer,
@@ -963,7 +963,7 @@ static int mp3Read(mp3DecodeData * data, ReplayGainInfo ** replayGainInfo)
 						       data->frameOffset[j]) ==
 				    0) {
 					data->outputPtr = data->outputBuffer;
-					clearOutputBuffer();
+					ob_clear();
 					data->currentFrame = j;
 				} else
 					dc.seekError = 1;
@@ -1029,7 +1029,7 @@ static int mp3_decode(InputStream * inStream)
 	}
 
 	initAudioFormatFromMp3DecodeData(&data, &(dc.audioFormat));
-	getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	dc.totalTime = data.totalTime;
 
@@ -1063,7 +1063,7 @@ static int mp3_decode(InputStream * inStream)
 	while (mp3Read(&data, &replayGainInfo) != DECODE_BREAK) ;
 	/* send last little bit if not dc.stop */
 	if (!dc.stop && data.outputPtr != data.outputBuffer && data.flush) {
-		sendDataToOutputBuffer(NULL,
+		ob_send(NULL,
 				       data.inStream->seekable,
 				       data.outputBuffer,
 				       data.outputPtr - data.outputBuffer,
@@ -1075,12 +1075,12 @@ static int mp3_decode(InputStream * inStream)
 		freeReplayGainInfo(replayGainInfo);
 
 	if (dc.seek && data.muteFrame == MUTEFRAME_SEEK) {
-		clearOutputBuffer();
+		ob_clear();
 		dc.seek = 0;
 		decoder_wakeup_player();
 	}
 
-	flushOutputBuffer();
+	ob_flush();
 	mp3DecodeDataFinalize(&data);
 
 	return 0;
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 1dd418b2d..7f13ca344 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -217,7 +217,7 @@ static int mp4_decode(InputStream * inStream)
 
 		if (seeking && seekPositionFound) {
 			seekPositionFound = 0;
-			clearOutputBuffer();
+			ob_clear();
 			seeking = 0;
 			dc.seek = 0;
 			decoder_wakeup_player();
@@ -255,7 +255,7 @@ static int mp4_decode(InputStream * inStream)
 			dc.audioFormat.sampleRate = scale;
 			dc.audioFormat.channels = frameInfo.channels;
 			getOutputAudioFormat(&(dc.audioFormat),
-					     &(cb.audioFormat));
+					     &(ob.audioFormat));
 			dc.state = DECODE_STATE_DECODE;
 		}
 
@@ -277,7 +277,7 @@ static int mp4_decode(InputStream * inStream)
 
 		sampleBuffer += offset * channels * 2;
 
-		sendDataToOutputBuffer(inStream, 1, sampleBuffer,
+		ob_send(inStream, 1, sampleBuffer,
 				       sampleBufferLen, file_time,
 				       bitRate, NULL);
 		if (dc.stop) {
@@ -295,11 +295,11 @@ static int mp4_decode(InputStream * inStream)
 		return -1;
 
 	if (dc.seek && seeking) {
-		clearOutputBuffer();
+		ob_clear();
 		dc.seek = 0;
 		decoder_wakeup_player();
 	}
-	flushOutputBuffer();
+	ob_flush();
 
 	return 0;
 }
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index 77ca07b30..1003f15d5 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -170,7 +170,7 @@ static int mpc_decode(InputStream * inStream)
 	dc.audioFormat.channels = info.channels;
 	dc.audioFormat.sampleRate = info.sample_freq;
 
-	getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	replayGainInfo = newReplayGainInfo();
 	replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -184,7 +184,7 @@ static int mpc_decode(InputStream * inStream)
 		if (dc.seek) {
 			samplePos = dc.seekWhere * dc.audioFormat.sampleRate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
-				clearOutputBuffer();
+				ob_clear();
 				s16 = (mpd_sint16 *) chunk;
 				chunkpos = 0;
 			} else
@@ -221,7 +221,7 @@ static int mpc_decode(InputStream * inStream)
 				bitRate = vbrUpdateBits *
 				    dc.audioFormat.sampleRate / 1152 / 1000;
 
-				sendDataToOutputBuffer(inStream,
+				ob_send(inStream,
 						       inStream->seekable,
 						       chunk, chunkpos,
 						       total_time,
@@ -243,12 +243,12 @@ static int mpc_decode(InputStream * inStream)
 		bitRate =
 		    vbrUpdateBits * dc.audioFormat.sampleRate / 1152 / 1000;
 
-		sendDataToOutputBuffer(NULL, inStream->seekable,
+		ob_send(NULL, inStream->seekable,
 				       chunk, chunkpos, total_time, bitRate,
 				       replayGainInfo);
 	}
 
-	flushOutputBuffer();
+	ob_flush();
 
 	freeReplayGainInfo(replayGainInfo);
 
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index 003b057d9..67259b626 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -362,7 +362,7 @@ static int oggflac_decode(InputStream * inStream)
 			    dc.audioFormat.sampleRate + 0.5;
 			if (OggFLAC__seekable_stream_decoder_seek_absolute
 			    (decoder, sampleToSeek)) {
-				clearOutputBuffer();
+				ob_clear();
 				data.time = ((float)sampleToSeek) /
 				    dc.audioFormat.sampleRate;
 				data.position = 0;
@@ -381,7 +381,7 @@ static int oggflac_decode(InputStream * inStream)
 	/* send last little bit */
 	if (data.chunk_length > 0 && !dc.stop) {
 		flacSendChunk(&data);
-		flushOutputBuffer();
+		ob_flush();
 	}
 
 fail:
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index eb44b5c6e..16040b388 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -275,7 +275,7 @@ static int oggvorbis_decode(InputStream * inStream)
 	while (1) {
 		if (dc.seek) {
 			if (0 == ov_time_seek_page(&vf, dc.seekWhere)) {
-				clearOutputBuffer();
+				ob_clear();
 				chunkpos = 0;
 			} else
 				dc.seekError = 1;
@@ -292,7 +292,7 @@ static int oggvorbis_decode(InputStream * inStream)
 			dc.audioFormat.sampleRate = vi->rate;
 			if (dc.state == DECODE_STATE_START) {
 				getOutputAudioFormat(&(dc.audioFormat),
-						     &(cb.audioFormat));
+						     &(ob.audioFormat));
 				dc.state = DECODE_STATE_DECODE;
 			}
 			comments = ov_comment(&vf, -1)->user_comments;
@@ -316,7 +316,7 @@ static int oggvorbis_decode(InputStream * inStream)
 			if ((test = ov_bitrate_instant(&vf)) > 0) {
 				bitRate = test / 1000;
 			}
-			sendDataToOutputBuffer(inStream,
+			ob_send(inStream,
 					       inStream->seekable,
 					       chunk, chunkpos,
 					       ov_pcm_tell(&vf) /
@@ -329,7 +329,7 @@ static int oggvorbis_decode(InputStream * inStream)
 	}
 
 	if (!dc.stop && chunkpos > 0) {
-		sendDataToOutputBuffer(NULL, inStream->seekable,
+		ob_send(NULL, inStream->seekable,
 				       chunk, chunkpos,
 				       ov_time_tell(&vf), bitRate,
 				       replayGainInfo);
@@ -340,7 +340,7 @@ static int oggvorbis_decode(InputStream * inStream)
 
 	ov_clear(&vf);
 
-	flushOutputBuffer();
+	ob_flush();
 
 	return 0;
 }
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index 13f10a1e9..6ab3d4d7a 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -166,7 +166,7 @@ static void wavpack_decode(WavpackContext *wpc, int canseek,
 
 	samplesreq = sizeof(chunk) / (4 * dc.audioFormat.channels);
 
-	getOutputAudioFormat(&(dc.audioFormat), &(cb.audioFormat));
+	getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
 
 	dc.totalTime = (float)allsamples / dc.audioFormat.sampleRate;
 	dc.state = DECODE_STATE_DECODE;
@@ -179,7 +179,7 @@ static void wavpack_decode(WavpackContext *wpc, int canseek,
 			if (canseek) {
 				int where;
 
-				clearOutputBuffer();
+				ob_clear();
 
 				where = dc.seekWhere *
 				        dc.audioFormat.sampleRate;
@@ -210,14 +210,14 @@ static void wavpack_decode(WavpackContext *wpc, int canseek,
 			format_samples(Bps, chunk,
 			               samplesgot * dc.audioFormat.channels);
 
-			sendDataToOutputBuffer(NULL, 0, chunk,
+			ob_send(NULL, 0, chunk,
 			                       samplesgot * outsamplesize,
 			                       file_time, bitrate,
 					       replayGainInfo);
 		}
 	} while (samplesgot == samplesreq);
 
-	flushOutputBuffer();
+	ob_flush();
 }
 
 static char *wavpack_tag(WavpackContext *wpc, char *key)
diff --git a/src/outputBuffer.c b/src/outputBuffer.c
index 1db523817..6732d0e7a 100644
--- a/src/outputBuffer.c
+++ b/src/outputBuffer.c
@@ -22,37 +22,37 @@
 #include "normalize.h"
 #include "playerData.h"
 
-void initOutputBuffer(unsigned int size)
+void ob_init(unsigned int size)
 {
 	assert(size > 0);
 
-	memset(&cb.convState, 0, sizeof(ConvState));
-	cb.chunks = xmalloc(size * sizeof(*cb.chunks));
-	cb.size = size;
-	cb.begin = 0;
-	cb.end = 0;
-	cb.chunks[0].chunkSize = 0;
+	memset(&ob.convState, 0, sizeof(ConvState));
+	ob.chunks = xmalloc(size * sizeof(*ob.chunks));
+	ob.size = size;
+	ob.begin = 0;
+	ob.end = 0;
+	ob.chunks[0].chunkSize = 0;
 }
 
-void output_buffer_free(void)
+void ob_free(void)
 {
-	assert(cb.chunks != NULL);
-	free(cb.chunks);
+	assert(ob.chunks != NULL);
+	free(ob.chunks);
 }
 
-void clearOutputBuffer(void)
+void ob_clear(void)
 {
-	cb.end = cb.begin;
-	cb.chunks[cb.end].chunkSize = 0;
+	ob.end = ob.begin;
+	ob.chunks[ob.end].chunkSize = 0;
 }
 
 /** return the index of the chunk after i */
 static inline unsigned successor(unsigned i)
 {
-	assert(i <= cb.size);
+	assert(i <= ob.size);
 
 	++i;
-	return i == cb.size ? 0 : i;
+	return i == ob.size ? 0 : i;
 }
 
 /**
@@ -61,13 +61,13 @@ static inline unsigned successor(unsigned i)
  */
 static void output_buffer_expand(unsigned i)
 {
-	int was_empty = outputBufferEmpty();
+	int was_empty = ob_is_empty();
 
-	assert(i == (cb.end + 1) % cb.size);
-	assert(i != cb.end);
+	assert(i == (ob.end + 1) % ob.size);
+	assert(i != ob.end);
 
-	cb.end = i;
-	cb.chunks[i].chunkSize = 0;
+	ob.end = i;
+	ob.chunks[i].chunkSize = 0;
 	if (was_empty)
 		/* if the buffer was empty, the player thread might be
 		   waiting for us; wake it up now that another decoded
@@ -75,13 +75,13 @@ static void output_buffer_expand(unsigned i)
 		decoder_wakeup_player();
 }
 
-void flushOutputBuffer(void)
+void ob_flush(void)
 {
-	OutputBufferChunk *chunk = outputBufferGetChunk(cb.end);
+	ob_chunk *chunk = ob_get_chunk(ob.end);
 
 	if (chunk->chunkSize > 0) {
-		unsigned int next = successor(cb.end);
-		if (next == cb.begin)
+		unsigned int next = successor(ob.end);
+		if (next == ob.begin)
 			/* all buffers are full; we have to wait for
 			   the player to free one, so don't flush
 			   right now */
@@ -91,54 +91,54 @@ void flushOutputBuffer(void)
 	}
 }
 
-int outputBufferEmpty(void)
+int ob_is_empty(void)
 {
-	return cb.begin == cb.end;
+	return ob.begin == ob.end;
 }
 
-void outputBufferShift(void)
+void ob_shift(void)
 {
-	assert(cb.begin != cb.end);
-	assert(cb.begin < cb.size);
+	assert(ob.begin != ob.end);
+	assert(ob.begin < ob.size);
 
-	cb.begin = successor(cb.begin);
+	ob.begin = successor(ob.begin);
 }
 
-unsigned int outputBufferRelative(const unsigned i)
+unsigned int ob_relative(const unsigned i)
 {
-	if (i >= cb.begin)
-		return i - cb.begin;
+	if (i >= ob.begin)
+		return i - ob.begin;
 	else
-		return i + cb.size - cb.begin;
+		return i + ob.size - ob.begin;
 }
 
-unsigned availableOutputBuffer(void)
+unsigned ob_available(void)
 {
-	return outputBufferRelative(cb.end);
+	return ob_relative(ob.end);
 }
 
-int outputBufferAbsolute(const unsigned relative)
+int ob_absolute(const unsigned relative)
 {
 	unsigned i, max;
 
-	max = cb.end;
-	if (max < cb.begin)
-		max += cb.size;
-	i = (unsigned)cb.begin + relative;
+	max = ob.end;
+	if (max < ob.begin)
+		max += ob.size;
+	i = (unsigned)ob.begin + relative;
 	if (i >= max)
 		return -1;
 
-	if (i >= cb.size)
-		i -= cb.size;
+	if (i >= ob.size)
+		i -= ob.size;
 
 	return (int)i;
 }
 
-OutputBufferChunk * outputBufferGetChunk(const unsigned i)
+ob_chunk * ob_get_chunk(const unsigned i)
 {
-	assert(i < cb.size);
+	assert(i < ob.size);
 
-	return &cb.chunks[i];
+	return &ob.chunks[i];
 }
 
 /**
@@ -154,14 +154,14 @@ static int tailChunk(InputStream * inStream,
 		     int seekable, float data_time, mpd_uint16 bitRate)
 {
 	unsigned int next;
-	OutputBufferChunk *chunk;
+	ob_chunk *chunk;
 
-	chunk = outputBufferGetChunk(cb.end);
+	chunk = ob_get_chunk(ob.end);
 	assert(chunk->chunkSize <= sizeof(chunk->data));
 	if (chunk->chunkSize == sizeof(chunk->data)) {
 		/* this chunk is full; allocate a new chunk */
-		next = successor(cb.end);
-		while (cb.begin == next) {
+		next = successor(ob.end);
+		while (ob.begin == next) {
 			/* all chunks are full of decoded data; wait
 			   for the player to free one */
 
@@ -183,7 +183,7 @@ static int tailChunk(InputStream * inStream,
 		}
 
 		output_buffer_expand(next);
-		chunk = outputBufferGetChunk(next);
+		chunk = ob_get_chunk(next);
 		assert(chunk->chunkSize == 0);
 	}
 
@@ -195,10 +195,10 @@ static int tailChunk(InputStream * inStream,
 		chunk->times = data_time;
 	}
 
-	return cb.end;
+	return ob.end;
 }
 
-int sendDataToOutputBuffer(InputStream * inStream,
+int ob_send(InputStream * inStream,
 			   int seekable, void *dataIn,
 			   size_t dataInLen, float data_time, mpd_uint16 bitRate,
 			   ReplayGainInfo * replayGainInfo)
@@ -208,14 +208,14 @@ int sendDataToOutputBuffer(InputStream * inStream,
 	size_t datalen;
 	static char *convBuffer;
 	static size_t convBufferLen;
-	OutputBufferChunk *chunk = NULL;
+	ob_chunk *chunk = NULL;
 
-	if (cmpAudioFormat(&(cb.audioFormat), &(dc.audioFormat)) == 0) {
+	if (cmpAudioFormat(&(ob.audioFormat), &(dc.audioFormat)) == 0) {
 		data = dataIn;
 		datalen = dataInLen;
 	} else {
 		datalen = pcm_sizeOfConvBuffer(&(dc.audioFormat), dataInLen,
-		                               &(cb.audioFormat));
+		                               &(ob.audioFormat));
 		if (datalen > convBufferLen) {
 			if (convBuffer != NULL)
 				free(convBuffer);
@@ -224,14 +224,14 @@ int sendDataToOutputBuffer(InputStream * inStream,
 		}
 		data = convBuffer;
 		datalen = pcm_convertAudioFormat(&(dc.audioFormat), dataIn,
-		                                 dataInLen, &(cb.audioFormat),
-		                                 data, &(cb.convState));
+		                                 dataInLen, &(ob.audioFormat),
+		                                 data, &(ob.convState));
 	}
 
 	if (replayGainInfo && (replayGainState != REPLAYGAIN_OFF))
-		doReplayGain(replayGainInfo, data, datalen, &cb.audioFormat);
+		doReplayGain(replayGainInfo, data, datalen, &ob.audioFormat);
 	else if (normalizationEnabled)
-		normalizeData(data, datalen, &cb.audioFormat);
+		normalizeData(data, datalen, &ob.audioFormat);
 
 	while (datalen) {
 		int chunk_index = tailChunk(inStream, seekable,
@@ -239,7 +239,7 @@ int sendDataToOutputBuffer(InputStream * inStream,
 		if (chunk_index < 0)
 			return chunk_index;
 
-		chunk = outputBufferGetChunk(chunk_index);
+		chunk = ob_get_chunk(chunk_index);
 
 		dataToSend = sizeof(chunk->data) - chunk->chunkSize;
 		if (dataToSend > datalen)
@@ -252,14 +252,14 @@ int sendDataToOutputBuffer(InputStream * inStream,
 	}
 
 	if (chunk != NULL && chunk->chunkSize == sizeof(chunk->data))
-		flushOutputBuffer();
+		ob_flush();
 
 	return 0;
 }
 
-void output_buffer_skip(unsigned num)
+void ob_skip(unsigned num)
 {
-	int i = outputBufferAbsolute(num);
+	int i = ob_absolute(num);
 	if (i >= 0)
-		cb.begin = i;
+		ob.begin = i;
 }
diff --git a/src/outputBuffer.h b/src/outputBuffer.h
index b0287192e..6e9d7c49f 100644
--- a/src/outputBuffer.h
+++ b/src/outputBuffer.h
@@ -36,14 +36,14 @@ typedef struct _OutputBufferChunk {
 	mpd_uint16 bitRate;
 	float times;
 	char data[CHUNK_SIZE];
-} OutputBufferChunk;
+} ob_chunk;
 
 /**
  * A ring set of buffers where the decoder appends data after the end,
  * and the player consumes data from the beginning.
  */
 typedef struct _OutputBuffer {
-	OutputBufferChunk *chunks;
+	ob_chunk *chunks;
 
 	unsigned int size;
 
@@ -57,45 +57,45 @@ typedef struct _OutputBuffer {
 	ConvState convState;
 } OutputBuffer;
 
-void initOutputBuffer(unsigned int size);
+void ob_init(unsigned int size);
 
-void output_buffer_free(void);
+void ob_free(void);
 
-void clearOutputBuffer(void);
+void ob_clear(void);
 
-void flushOutputBuffer(void);
+void ob_flush(void);
 
 /** is the buffer empty? */
-int outputBufferEmpty(void);
+int ob_is_empty(void);
 
-void outputBufferShift(void);
+void ob_shift(void);
 
 /**
  * what is the position of the specified chunk number, relative to
  * the first chunk in use?
  */
-unsigned int outputBufferRelative(const unsigned i);
+unsigned int ob_relative(const unsigned i);
 
 /** determine the number of decoded chunks */
-unsigned availableOutputBuffer(void);
+unsigned ob_available(void);
 
 /**
  * Get the absolute index of the nth used chunk after the first one.
  * Returns -1 if there is no such chunk.
  */
-int outputBufferAbsolute(const unsigned relative);
+int ob_absolute(const unsigned relative);
 
-OutputBufferChunk * outputBufferGetChunk(const unsigned i);
+ob_chunk * ob_get_chunk(const unsigned i);
 
 /* we send inStream for buffering the inputStream while waiting to
    send the next chunk */
-int sendDataToOutputBuffer(InputStream * inStream,
+int ob_send(InputStream * inStream,
 			   int seekable,
 			   void *data,
 			   size_t datalen,
 			   float data_time,
 			   mpd_uint16 bitRate, ReplayGainInfo * replayGainInfo);
 
-void output_buffer_skip(unsigned num);
+void ob_skip(unsigned num);
 
 #endif
diff --git a/src/playerData.c b/src/playerData.c
index 5ac7c4785..56dee348c 100644
--- a/src/playerData.c
+++ b/src/playerData.c
@@ -29,7 +29,7 @@ unsigned int buffered_before_play;
 static PlayerData playerData_pd;
 PlayerControl pc;
 DecoderControl dc;
-OutputBuffer cb; /* rename this to 'ob' */
+OutputBuffer ob;
 
 void initPlayerData(void)
 {
@@ -77,7 +77,7 @@ void initPlayerData(void)
 
 	playerData_pd.audioDeviceStates = xmalloc(device_array_size);
 
-	initOutputBuffer(buffered_chunks);
+	ob_init(buffered_chunks);
 
 	notifyInit(&pc.notify);
 	pc.error = PLAYER_ERROR_NOERROR;
@@ -104,6 +104,6 @@ void freePlayerData(void)
 	 * access playerData_pd and we need to keep it available for them */
 	waitpid(-1, NULL, 0);
 
-	output_buffer_free();
+	ob_free();
 	free(playerData_pd.audioDeviceStates);
 }
diff --git a/src/playerData.h b/src/playerData.h
index 80423717d..fccc758da 100644
--- a/src/playerData.h
+++ b/src/playerData.h
@@ -28,7 +28,7 @@
 extern unsigned int buffered_before_play;
 extern PlayerControl pc;
 extern DecoderControl dc;
-extern OutputBuffer cb; /* rename this to 'ob' */
+extern OutputBuffer ob;
 
 typedef struct _PlayerData {
 	mpd_uint8 *audioDeviceStates;