audio_format: renamed sampleRate to sample_rate

src/audio.c

@@ -137,16 +137,16 @@ int parseAudioConfig(struct audio_format *audioFormat, char *conf)
 
 	memset(audioFormat, 0, sizeof(*audioFormat));
 
-	audioFormat->sampleRate = strtol(conf, &test, 10);
+	audioFormat->sample_rate = strtol(conf, &test, 10);
 
 	if (*test != ':') {
 		ERROR("error parsing audio output format: %s\n", conf);
 		return -1;
 	}
 
-	if (audioFormat->sampleRate <= 0) {
-		ERROR("sample rate %i is not >= 0\n",
-		      (int)audioFormat->sampleRate);
+	if (audioFormat->sample_rate <= 0) {
+		ERROR("sample rate %u is not >= 0\n",
+		      audioFormat->sample_rate);
 		return -1;
 	}
 
@@ -315,7 +315,7 @@ static int flushAudioBuffer(void)
 
 static size_t audio_buffer_size(const struct audio_format *af)
 {
-	return (af->bits >> 3) * af->channels * (af->sampleRate >> 5);
+	return (af->bits >> 3) * af->channels * (af->sample_rate >> 5);
 }
 
 static void audio_buffer_resize(size_t size)

src/audioOutputs/audioOutput_alsa.c

@@ -142,7 +142,7 @@ static int alsa_openDevice(void *data, struct audio_format *audioFormat)
 	snd_pcm_format_t bitformat;
 	snd_pcm_hw_params_t *hwparams;
 	snd_pcm_sw_params_t *swparams;
-	unsigned int sampleRate = audioFormat->sampleRate;
+	unsigned int sample_rate = audioFormat->sample_rate;
 	unsigned int channels = audioFormat->channels;
 	snd_pcm_uframes_t alsa_buffer_size;
 	snd_pcm_uframes_t alsa_period_size;
@@ -217,13 +217,13 @@ configure_hw:
 	audioFormat->channels = (int8_t)channels;
 
 	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
-					      &sampleRate, NULL);
-	if (err < 0 || sampleRate == 0) {
-		ERROR("ALSA device \"%s\" does not support %i Hz audio\n",
-		      ad->device, (int)audioFormat->sampleRate);
+					      &sample_rate, NULL);
+	if (err < 0 || sample_rate == 0) {
+		ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
+		      ad->device, audioFormat->sample_rate);
 		goto fail;
 	}
-	audioFormat->sampleRate = sampleRate;
+	audioFormat->sample_rate = sample_rate;
 
 	buffer_time = ad->buffer_time;
 	cmd = "snd_pcm_hw_params_set_buffer_time_near";
@@ -291,8 +291,8 @@ configure_hw:
 	ad->sampleSize = audio_format_sample_size(audioFormat) * audioFormat->channels;
 
 	DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
-	      "%i Hz\n", ad->device, audioFormat->bits,
-	      channels, sampleRate);
+	      "%u Hz\n", ad->device, audioFormat->bits,
+	      channels, sample_rate);
 
 	return 0;
 

src/audioOutputs/audioOutput_ao.c

@@ -182,7 +182,7 @@ static int audioOutputAo_openDevice(void *data,
 	}
 
 	format.bits = audio_format->bits;
-	format.rate = audio_format->sampleRate;
+	format.rate = audio_format->sample_rate;
 	format.byte_format = AO_FMT_NATIVE;
 	format.channels = audio_format->channels;
 

src/audioOutputs/audioOutput_jack.c

@@ -126,7 +126,7 @@ static int srate(mpd_unused jack_nframes_t rate, void *data)
 	JackData *jd = (JackData *)data;
 	struct audio_format *audioFormat = jd->audio_format;
 
- 	audioFormat->sampleRate = (int)jack_get_sample_rate(jd->client);
+	audioFormat->sample_rate = (int)jack_get_sample_rate(jd->client);
 
 	return 0;
 }
@@ -188,13 +188,13 @@ static void shutdown_callback(void *arg)
 
 static void set_audioformat(JackData *jd, struct audio_format *audioFormat)
 {
-	audioFormat->sampleRate = (int) jack_get_sample_rate(jd->client);
-	DEBUG("samplerate = %d\n", audioFormat->sampleRate);
+	audioFormat->sample_rate = jack_get_sample_rate(jd->client);
+	DEBUG("samplerate = %u\n", audioFormat->sample_rate);
 	audioFormat->channels = 2;
 	audioFormat->bits = 16;
 	jd->bps = audioFormat->channels
 		* sizeof(jack_default_audio_sample_t)
-		* audioFormat->sampleRate;
+		* audioFormat->sample_rate;
 }
 
 static void error_callback(const char *msg)

src/audioOutputs/audioOutput_mvp.c

@@ -202,11 +202,11 @@ static int mvp_openDevice(struct audio_output *audioOutput,
 		return -1;
 	}
 #ifdef WORDS_BIGENDIAN
-	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 0,
-			 audioFormat->bits);
+	mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
+			 0, audioFormat->bits);
 #else
-	mvp_setPcmParams(md, audioFormat->sampleRate, audioFormat->channels, 1,
-			 audioFormat->bits);
+	mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
+			 1, audioFormat->bits);
 #endif
 	return 0;
 }

src/audioOutputs/audioOutput_oss.c

@@ -487,14 +487,14 @@ static int oss_openDevice(void *data,
 	OssData *od = data;
 
 	od->channels = (int8_t)audioFormat->channels;
-	od->sampleRate = audioFormat->sampleRate;
+	od->sampleRate = audioFormat->sample_rate;
 	od->bits = (int8_t)audioFormat->bits;
 
 	if ((ret = oss_open(od)) < 0)
 		return ret;
 
 	audioFormat->channels = od->channels;
-	audioFormat->sampleRate = od->sampleRate;
+	audioFormat->sample_rate = od->sampleRate;
 	audioFormat->bits = od->bits;
 
 	DEBUG("oss device \"%s\" will be playing %i bit %i channel audio at "

src/audioOutputs/audioOutput_osx.c

@@ -259,7 +259,7 @@ static int osx_openDevice(struct audio_output *audioOutput,
 		return -1;
 	}
 
-	streamDesc.mSampleRate = audioFormat->sampleRate;
+	streamDesc.mSampleRate = audioFormat->sample_rate;
 	streamDesc.mFormatID = kAudioFormatLinearPCM;
 	streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
 #ifdef WORDS_BIGENDIAN
@@ -283,7 +283,7 @@ static int osx_openDevice(struct audio_output *audioOutput,
 	}
 
 	/* create a buffer of 1s */
-	od->bufferSize = (audioFormat->sampleRate) *
+	od->bufferSize = (audioFormat->sample_rate) *
 	    (audioFormat->bits >> 3) * (audioFormat->channels);
 	od->buffer = xrealloc(od->buffer, od->bufferSize);
 

src/audioOutputs/audioOutput_pulse.c

@@ -138,7 +138,7 @@ static int pulse_openDevice(void *data,
 	}
 
 	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = audioFormat->sampleRate;
+	ss.rate = audioFormat->sample_rate;
 	ss.channels = audioFormat->channels;
 
 	pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
@@ -159,7 +159,7 @@ static int pulse_openDevice(void *data,
 	      "channel audio at %i Hz\n",
 	      audio_output_get_name(pd->ao),
 	      audioFormat->bits,
-	      audioFormat->channels, audioFormat->sampleRate);
+	      audioFormat->channels, audioFormat->sample_rate);
 
 	return 0;
 }

src/audioOutputs/audioOutput_shout.c

@@ -255,7 +255,7 @@ static void *my_shout_init_driver(struct audio_output *audio_output,
 		snprintf(temp, sizeof(temp), "%u", sd->audio_format.channels);
 		shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
 
-		snprintf(temp, sizeof(temp), "%d", sd->audio_format.sampleRate);
+		snprintf(temp, sizeof(temp), "%u", sd->audio_format.sample_rate);
 
 		shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
 

src/audioOutputs/audioOutput_shout_mp3.c

@@ -93,7 +93,7 @@ static int shout_mp3_encoder_init_encoder(struct shout_data *sd)
 	}
 
 	if (0 != lame_set_in_samplerate(ld->gfp,
-					sd->audio_format.sampleRate)) {
+					sd->audio_format.sample_rate)) {
 		ERROR("error setting lame sample rate\n");
 		return -1;
 	}

src/audioOutputs/audioOutput_shout_ogg.c

@@ -187,7 +187,7 @@ static int reinit_encoder(struct shout_data *sd)
 	if (sd->quality >= -1.0) {
 		if (0 != vorbis_encode_init_vbr(&od->vi,
 						sd->audio_format.channels,
-						sd->audio_format.sampleRate,
+						sd->audio_format.sample_rate,
 						sd->quality * 0.1)) {
 			ERROR("error initializing vorbis vbr\n");
 			vorbis_info_clear(&od->vi);
@@ -196,7 +196,7 @@ static int reinit_encoder(struct shout_data *sd)
 	} else {
 		if (0 != vorbis_encode_init(&od->vi,
 					    sd->audio_format.channels,
-					    sd->audio_format.sampleRate, -1.0,
+					    sd->audio_format.sample_rate, -1.0,
 					    sd->bitrate * 1000, -1.0)) {
 			ERROR("error initializing vorbis encoder\n");
 			vorbis_info_clear(&od->vi);

src/audio_format.h

@@ -23,27 +23,27 @@
 #include <stdbool.h>
 
 struct audio_format {
-	uint32_t sampleRate;
+	uint32_t sample_rate;
 	uint8_t bits;
 	uint8_t channels;
 };
 
 static inline void audio_format_clear(struct audio_format *af)
 {
-	af->sampleRate = 0;
+	af->sample_rate = 0;
 	af->bits = 0;
 	af->channels = 0;
 }
 
 static inline bool audio_format_defined(const struct audio_format *af)
 {
-	return af->sampleRate != 0;
+	return af->sample_rate != 0;
 }
 
 static inline bool audio_format_equals(const struct audio_format *a,
 				       const struct audio_format *b)
 {
-	return a->sampleRate == b->sampleRate &&
+	return a->sample_rate == b->sample_rate &&
 		a->bits == b->bits &&
 		a->channels == b->channels;
 }
@@ -63,7 +63,7 @@ static inline unsigned audio_format_sample_size(const struct audio_format *af)
 
 static inline double audio_format_time_to_size(const struct audio_format *af)
 {
-	return af->sampleRate * af->channels * audio_format_sample_size(af);
+	return af->sample_rate * af->channels * audio_format_sample_size(af);
 }
 
 static inline double audioFormatSizeToTime(const struct audio_format *af)

src/crossfade.c

@@ -37,7 +37,7 @@ unsigned cross_fade_calc(float duration, float total_time,
 	assert(duration > 0);
 	assert(af->bits > 0);
 	assert(af->channels > 0);
-	assert(af->sampleRate > 0);
+	assert(af->sample_rate > 0);
 
 	chunks = audio_format_time_to_size(af) / CHUNK_SIZE;
 	chunks = (chunks * duration + 0.5);

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;

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 */
@@ -158,7 +158,7 @@ static void adtsParse(AacBuffer * b, float *length)
 		frameLength = adts_find_frame(b);
 		if (frameLength > 0) {
 			if (frames == 0) {
-				sampleRate = adtsSampleRates[(b->
+				sample_rate = adtsSampleRates[(b->
 							       buffer[2] & 0x3c)
 							      >> 2];
 			}
@@ -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;

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);
 

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

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);
 

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);
 }
 

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);

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,

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

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)

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);

src/pcm_utils.c

@@ -503,13 +503,13 @@ size_t pcm_convertAudioFormat(const struct audio_format *inFormat,
 			exit(EXIT_FAILURE);
 	}
 
-	if (inFormat->sampleRate == outFormat->sampleRate) {
+	if (inFormat->sample_rate == outFormat->sample_rate) {
 		assert(outSize >= len);
 		memcpy(outBuffer, buf, len);
 	} else {
 		len = pcm_convertSampleRate(outFormat->channels,
-		                            inFormat->sampleRate, buf, len,
-		                            outFormat->sampleRate, outBuffer,
+		                            inFormat->sample_rate, buf, len,
+		                            outFormat->sample_rate, outBuffer,
 		                            outSize, convState);
 		if (len == 0)
 			exit(EXIT_FAILURE);
@@ -521,8 +521,8 @@ size_t pcm_convertAudioFormat(const struct audio_format *inFormat,
 size_t pcm_sizeOfConvBuffer(const struct audio_format *inFormat, size_t inSize,
                             const struct audio_format *outFormat)
 {
-	const double ratio = (double)outFormat->sampleRate /
-	                     (double)inFormat->sampleRate;
+	const double ratio = (double)outFormat->sample_rate /
+	                     (double)inFormat->sample_rate;
 	const int shift = 2 * outFormat->channels;
 	size_t outSize = inSize;
 

src/player_thread.c

@@ -253,7 +253,7 @@ static void do_play(void)
 					closeAudioDevice();
 				}
 				pc.totalTime = dc.totalTime;
-				pc.sampleRate = dc.audioFormat.sampleRate;
+				pc.sampleRate = dc.audioFormat.sample_rate;
 				pc.bits = dc.audioFormat.bits;
 				pc.channels = dc.audioFormat.channels;
 				sizeToTime = audioFormatSizeToTime(&ob.audioFormat);

src/timer.c

@@ -40,7 +40,7 @@ Timer *timer_new(const struct audio_format *af)
 	timer = xmalloc(sizeof(Timer));
 	timer->time = 0;
 	timer->started = 0;
-	timer->rate = af->sampleRate * (af->bits / CHAR_BIT) * af->channels;
+	timer->rate = af->sample_rate * (af->bits / CHAR_BIT) * af->channels;
 
 	return timer;
 }