audio_format: renamed sampleRate to sample_rate

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

10 files changed, 27 insertions, 27 deletions

diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 83bd9c256..30ad449f3 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/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;
 
diff --git a/src/audioOutputs/audioOutput_ao.c b/src/audioOutputs/audioOutput_ao.c
index b91895bde..e731f972a 100644
--- a/src/audioOutputs/audioOutput_ao.c
+++ b/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;
 
diff --git a/src/audioOutputs/audioOutput_jack.c b/src/audioOutputs/audioOutput_jack.c
index f26dfcf7a..8a2cb6cdc 100644
--- a/src/audioOutputs/audioOutput_jack.c
+++ b/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)
diff --git a/src/audioOutputs/audioOutput_mvp.c b/src/audioOutputs/audioOutput_mvp.c
index 59f43a4fd..00b069c3d 100644
--- a/src/audioOutputs/audioOutput_mvp.c
+++ b/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;
 }
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
index 487e9a75d..8dddf3be7 100644
--- a/src/audioOutputs/audioOutput_oss.c
+++ b/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 "
diff --git a/src/audioOutputs/audioOutput_osx.c b/src/audioOutputs/audioOutput_osx.c
index 9071ed6c9..1fc0a5d9e 100644
--- a/src/audioOutputs/audioOutput_osx.c
+++ b/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);
 
diff --git a/src/audioOutputs/audioOutput_pulse.c b/src/audioOutputs/audioOutput_pulse.c
index 38014c8f0..93a1d8b37 100644
--- a/src/audioOutputs/audioOutput_pulse.c
+++ b/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;
 }
diff --git a/src/audioOutputs/audioOutput_shout.c b/src/audioOutputs/audioOutput_shout.c
index 34327573c..00c4eb059 100644
--- a/src/audioOutputs/audioOutput_shout.c
+++ b/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);
 
diff --git a/src/audioOutputs/audioOutput_shout_mp3.c b/src/audioOutputs/audioOutput_shout_mp3.c
index c54632b15..722079b29 100644
--- a/src/audioOutputs/audioOutput_shout_mp3.c
+++ b/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;
 	}
diff --git a/src/audioOutputs/audioOutput_shout_ogg.c b/src/audioOutputs/audioOutput_shout_ogg.c
index 14747c324..5983b4d89 100644
--- a/src/audioOutputs/audioOutput_shout_ogg.c
+++ b/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);