From de2cb3f37568e7680549057f8d7b6d748c388480 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Fri, 10 Oct 2008 14:40:54 +0200 Subject: audio_format: renamed sampleRate to sample_rate The last bit of CamelCase in audio_format.h. Additionally, rename a bunch of local variables. --- src/audioOutputs/audioOutput_alsa.c | 16 ++++++++-------- src/audioOutputs/audioOutput_ao.c | 2 +- src/audioOutputs/audioOutput_jack.c | 8 ++++---- src/audioOutputs/audioOutput_mvp.c | 8 ++++---- src/audioOutputs/audioOutput_oss.c | 4 ++-- src/audioOutputs/audioOutput_osx.c | 4 ++-- src/audioOutputs/audioOutput_pulse.c | 4 ++-- src/audioOutputs/audioOutput_shout.c | 2 +- src/audioOutputs/audioOutput_shout_mp3.c | 2 +- src/audioOutputs/audioOutput_shout_ogg.c | 4 ++-- 10 files changed, 27 insertions(+), 27 deletions(-) (limited to 'src/audioOutputs') 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); -- cgit v1.2.3