audio_format: renamed sampleRate to sample_rate

The last bit of CamelCase in audio_format.h.  Additionally, rename a
bunch of local variables.

11 files changed, 51 insertions, 52 deletions

diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 22d8774a3..f24e20531 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -162,7 +162,7 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
 		data->audio_format.bits = (int8_t)si->bits_per_sample;
-		data->audio_format.sampleRate = si->sample_rate;
+		data->audio_format.sample_rate = si->sample_rate;
 		data->audio_format.channels = (int8_t)si->channels;
 		data->total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index a96623e1b..e9b2d7476 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -148,7 +148,7 @@ static size_t adts_find_frame(AacBuffer * b)
 static void adtsParse(AacBuffer * b, float *length)
 {
 	unsigned int frames, frameLength;
-	int sampleRate = 0;
+	int sample_rate = 0;
 	float framesPerSec;
 
 	/* Read all frames to ensure correct time and bitrate */
@@ -158,9 +158,9 @@ static void adtsParse(AacBuffer * b, float *length)
 		frameLength = adts_find_frame(b);
 		if (frameLength > 0) {
 			if (frames == 0) {
-				sampleRate = adtsSampleRates[(b->
-							      buffer[2] & 0x3c)
-							     >> 2];
+				sample_rate = adtsSampleRates[(b->
+							       buffer[2] & 0x3c)
+							      >> 2];
 			}
 
 			if (frameLength > b->bytesIntoBuffer)
@@ -171,7 +171,7 @@ static void adtsParse(AacBuffer * b, float *length)
 			break;
 	}
 
-	framesPerSec = (float)sampleRate / 1024.0;
+	framesPerSec = (float)sample_rate / 1024.0;
 	if (framesPerSec != 0)
 		*length = (float)frames / framesPerSec;
 }
@@ -253,7 +253,7 @@ static float getAacFloatTotalTime(char *file)
 	float length;
 	faacDecHandle decoder;
 	faacDecConfigurationPtr config;
-	uint32_t sampleRate;
+	uint32_t sample_rate;
 	unsigned char channels;
 	InputStream inStream;
 	long bread;
@@ -274,11 +274,11 @@ static float getAacFloatTotalTime(char *file)
 		fillAacBuffer(&b);
 #ifdef HAVE_FAAD_BUFLEN_FUNCS
 		bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
-				    &sampleRate, &channels);
+				    &sample_rate, &channels);
 #else
-		bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+		bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
 #endif
-		if (bread >= 0 && sampleRate > 0 && channels > 0)
+		if (bread >= 0 && sample_rate > 0 && channels > 0)
 			length = 0;
 
 		faacDecClose(decoder);
@@ -312,7 +312,7 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
 	faacDecConfigurationPtr config;
 	long bread;
 	struct audio_format audio_format;
-	uint32_t sampleRate;
+	uint32_t sample_rate;
 	unsigned char channels;
 	unsigned int sampleCount;
 	char *sampleBuffer;
@@ -346,9 +346,9 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
 
 #ifdef HAVE_FAAD_BUFLEN_FUNCS
 	bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
-			    &sampleRate, &channels);
+			    &sample_rate, &channels);
 #else
-	bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+	bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
 #endif
 	if (bread < 0) {
 		ERROR("Error not a AAC stream.\n");
@@ -386,12 +386,12 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
 			break;
 		}
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
-		sampleRate = frameInfo.samplerate;
+		sample_rate = frameInfo.samplerate;
 #endif
 
 		if (!initialized) {
 			audio_format.channels = frameInfo.channels;
-			audio_format.sampleRate = sampleRate;
+			audio_format.sample_rate = sample_rate;
 			decoder_initialized(mpd_decoder, &audio_format, totalTime);
 			initialized = 1;
 		}
@@ -402,11 +402,11 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
 
 		if (sampleCount > 0) {
 			bitRate = frameInfo.bytesconsumed * 8.0 *
-			    frameInfo.channels * sampleRate /
+			    frameInfo.channels * sample_rate /
 			    frameInfo.samples / 1000 + 0.5;
 			file_time +=
 			    (float)(frameInfo.samples) / frameInfo.channels /
-			    sampleRate;
+			    sample_rate;
 		}
 
 		sampleBufferLen = sampleCount * 2;
@@ -446,7 +446,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 	faacDecConfigurationPtr config;
 	long bread;
 	struct audio_format audio_format;
-	uint32_t sampleRate;
+	uint32_t sample_rate;
 	unsigned char channels;
 	unsigned int sampleCount;
 	char *sampleBuffer;
@@ -484,9 +484,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 
 #ifdef HAVE_FAAD_BUFLEN_FUNCS
 	bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
-			    &sampleRate, &channels);
+			    &sample_rate, &channels);
 #else
-	bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+	bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
 #endif
 	if (bread < 0) {
 		ERROR("Error not a AAC stream.\n");
@@ -522,12 +522,12 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 			break;
 		}
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
-		sampleRate = frameInfo.samplerate;
+		sample_rate = frameInfo.samplerate;
 #endif
 
 		if (!initialized) {
 			audio_format.channels = frameInfo.channels;
-			audio_format.sampleRate = sampleRate;
+			audio_format.sample_rate = sample_rate;
 			decoder_initialized(mpd_decoder, &audio_format,
 					    totalTime);
 			initialized = 1;
@@ -539,11 +539,11 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
 
 		if (sampleCount > 0) {
 			bitRate = frameInfo.bytesconsumed * 8.0 *
-			    frameInfo.channels * sampleRate /
+			    frameInfo.channels * sample_rate /
 			    frameInfo.samples / 1000 + 0.5;
 			file_time +=
 			    (float)(frameInfo.samples) / frameInfo.channels /
-			    sampleRate;
+			    sample_rate;
 		}
 
 		sampleBufferLen = sampleCount * 2;
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 421cdf354..4c08074c4 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -71,14 +71,14 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	                         AF_SAMPFMT_TWOSCOMP, 16);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
 	audio_format.bits = (uint8_t)bits;
-	audio_format.sampleRate =
+	audio_format.sample_rate =
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
 	audio_format.channels =
 	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
-	total_time = ((float)frame_count / (float)audio_format.sampleRate);
+	total_time = ((float)frame_count / (float)audio_format.sample_rate);
 
 	bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
 
@@ -97,7 +97,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
 			decoder_clear(decoder);
 			current = decoder_seek_where(decoder) *
-				audio_format.sampleRate;
+				audio_format.sample_rate;
 			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
 			decoder_command_finished(decoder);
 		}
@@ -110,7 +110,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 		current += ret;
 		decoder_data(decoder, NULL, 1,
 			     chunk, ret * fs,
-			     (float)current / (float)audio_format.sampleRate,
+			     (float)current / (float)audio_format.sample_rate,
 			     bitRate, NULL);
 	} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
 
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index cd8a8efd3..2f3ec88d9 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -350,11 +350,11 @@ static int flac_decode_internal(struct decoder * decoder,
 			break;
 		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
 			FLAC__uint64 sampleToSeek = decoder_seek_where(decoder) *
-			    data.audio_format.sampleRate + 0.5;
+			    data.audio_format.sample_rate + 0.5;
 			if (flac_seek_absolute(flacDec, sampleToSeek)) {
 				decoder_clear(decoder);
 				data.time = ((float)sampleToSeek) /
-				    data.audio_format.sampleRate;
+				    data.audio_format.sample_rate;
 				data.position = 0;
 				decoder_command_finished(decoder);
 			} else
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 9ae9cef16..98bd67f0f 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -186,12 +186,12 @@ static int mod_decode(struct decoder * decoder, char *path)
 	}
 
 	audio_format.bits = 16;
-	audio_format.sampleRate = 44100;
+	audio_format.sample_rate = 44100;
 	audio_format.channels = 2;
 
 	secPerByte =
 	    1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
-		   (float)audio_format.sampleRate);
+		   (float)audio_format.sample_rate);
 
 	decoder_initialized(decoder, &audio_format, 0);
 
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 60e09a1bb..2990de1ac 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -1030,7 +1030,7 @@ static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
 					     struct audio_format * af)
 {
 	af->bits = 16;
-	af->sampleRate = (data->frame).header.samplerate;
+	af->sample_rate = (data->frame).header.samplerate;
 	af->channels = MAD_NCHANNELS(&(data->frame).header);
 }
 
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 6a2d167b2..d284313d4 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -92,7 +92,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	struct audio_format audio_format;
 	unsigned char *mp4Buffer;
 	unsigned int mp4BufferSize;
-	uint32_t sampleRate;
+	uint32_t sample_rate;
 	unsigned char channels;
 	long sampleId;
 	long numSamples;
@@ -149,7 +149,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
 
 	if (faacDecInit2
-	    (decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels) < 0) {
+	    (decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) {
 		ERROR("Error not a AAC stream.\n");
 		faacDecClose(decoder);
 		mp4ff_close(mp4fh);
@@ -157,7 +157,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		return -1;
 	}
 
-	audio_format.sampleRate = sampleRate;
+	audio_format.sample_rate = sample_rate;
 	audio_format.channels = channels;
 	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
 	scale = mp4ff_time_scale(mp4fh, track);
@@ -255,7 +255,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
 			scale = frameInfo.samplerate;
 #endif
-			audio_format.sampleRate = scale;
+			audio_format.sample_rate = scale;
 			audio_format.channels = frameInfo.channels;
 			decoder_initialized(mpd_decoder, &audio_format,
 					    total_time);
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index ca37333d3..f74dc8ddc 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -154,7 +154,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 
 	audio_format.bits = 16;
 	audio_format.channels = info.channels;
-	audio_format.sampleRate = info.sample_freq;
+	audio_format.sample_rate = info.sample_freq;
 
 	replayGainInfo = newReplayGainInfo();
 	replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -168,7 +168,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	while (!eof) {
 		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
 			samplePos = decoder_seek_where(mpd_decoder) *
-				audio_format.sampleRate;
+				audio_format.sample_rate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
 				decoder_clear(mpd_decoder);
 				s16 = (int16_t *) chunk;
@@ -201,10 +201,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 
 			if (chunkpos >= MPC_CHUNK_SIZE) {
 				total_time = ((float)samplePos) /
-				    audio_format.sampleRate;
+				    audio_format.sample_rate;
 
 				bitRate = vbrUpdateBits *
-				    audio_format.sampleRate / 1152 / 1000;
+				    audio_format.sample_rate / 1152 / 1000;
 
 				decoder_data(mpd_decoder, inStream,
 					     inStream->seekable,
@@ -224,10 +224,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 
 	if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP &&
 	    chunkpos > 0) {
-		total_time = ((float)samplePos) / audio_format.sampleRate;
+		total_time = ((float)samplePos) / audio_format.sample_rate;
 
 		bitRate =
-		    vbrUpdateBits * audio_format.sampleRate / 1152 / 1000;
+		    vbrUpdateBits * audio_format.sample_rate / 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 3a2db5c03..53f233e0c 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -316,12 +316,12 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
 		}
 		if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
 			FLAC__uint64 sampleToSeek = decoder_seek_where(mpd_decoder) *
-			    data.audio_format.sampleRate + 0.5;
+			    data.audio_format.sample_rate + 0.5;
 			if (OggFLAC__seekable_stream_decoder_seek_absolute
 			    (decoder, sampleToSeek)) {
 				decoder_clear(mpd_decoder);
 				data.time = ((float)sampleToSeek) /
-				    data.audio_format.sampleRate;
+				    data.audio_format.sample_rate;
 				data.position = 0;
 				decoder_command_finished(mpd_decoder);
 			} else
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index 0e1d523b9..bf2448605 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -278,7 +278,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 			/*printf("new song!\n"); */
 			vorbis_info *vi = ov_info(&vf, -1);
 			audio_format.channels = vi->channels;
-			audio_format.sampleRate = vi->rate;
+			audio_format.sample_rate = vi->rate;
 			if (!initialized) {
 				float total_time = ov_time_total(&vf, -1);
 				if (total_time < 0)
@@ -311,7 +311,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
 			decoder_data(decoder, inStream,
 				     inStream->seekable,
 				     chunk, chunkpos,
-				     ov_pcm_tell(&vf) / audio_format.sampleRate,
+				     ov_pcm_tell(&vf) / audio_format.sample_rate,
 				     bitRate, replayGainInfo);
 			chunkpos = 0;
 			if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index af7c3a2f3..3e99980bd 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -140,7 +140,7 @@ static void wavpack_decode(struct decoder * decoder,
 	int position, outsamplesize;
 	int Bps;
 
-	audio_format.sampleRate = WavpackGetSampleRate(wpc);
+	audio_format.sample_rate = WavpackGetSampleRate(wpc);
 	audio_format.channels = WavpackGetReducedChannels(wpc);
 	audio_format.bits = WavpackGetBitsPerSample(wpc);
 
@@ -168,7 +168,7 @@ static void wavpack_decode(struct decoder * decoder,
 	samplesreq = sizeof(chunk) / (4 * audio_format.channels);
 
 	decoder_initialized(decoder, &audio_format,
-			    (float)allsamples / audio_format.sampleRate);
+			    (float)allsamples / audio_format.sample_rate);
 
 	position = 0;
 
@@ -180,7 +180,7 @@ static void wavpack_decode(struct decoder * decoder,
 				decoder_clear(decoder);
 
 				where = decoder_seek_where(decoder) *
-				        audio_format.sampleRate;
+					audio_format.sample_rate;
 				if (WavpackSeekSample(wpc, where)) {
 					position = where;
 					decoder_command_finished(decoder);
@@ -200,8 +200,7 @@ static void wavpack_decode(struct decoder * decoder,
 			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
 			              1000 + 0.5);
 			position += samplesgot;
-			file_time = (float)position /
-			            audio_format.sampleRate;
+			file_time = (float)position / audio_format.sample_rate;
 
 			format_samples(Bps, chunk,
 			               samplesgot * audio_format.channels);