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

12 files changed, 42 insertions, 42 deletions

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)