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audio_format: renamed sampleRate to sample_rate

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The last bit of CamelCase in audio_format.h.  Additionally, rename a
bunch of local variables.
This commit is contained in:
Max Kellermann 2008-10-10 14:40:54 +02:00
parent 6101dc6c76
commit de2cb3f375
27 changed files with 96 additions and 97 deletions
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10
src/audio.c
View File

@ -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)

16
src/audioOutputs/audioOutput_alsa.c
View File

@ -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;

2
src/audioOutputs/audioOutput_ao.c
View File

@ -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;

8
src/audioOutputs/audioOutput_jack.c
View File

@ -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)

8
src/audioOutputs/audioOutput_mvp.c
View File

@ -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;
}

4
src/audioOutputs/audioOutput_oss.c
View File

@ -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 "

4
src/audioOutputs/audioOutput_osx.c
View File

@ -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);

4
src/audioOutputs/audioOutput_pulse.c
View File

@ -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;
}

2
src/audioOutputs/audioOutput_shout.c
View File

@ -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);

2
src/audioOutputs/audioOutput_shout_mp3.c
View File

@ -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;
}

4
src/audioOutputs/audioOutput_shout_ogg.c
View File

@ -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);

10
src/audio_format.h
View File

@ -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)

2
src/crossfade.c
View File

@ -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);

2
src/inputPlugins/_flac_common.c
View File

@ -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;

46
src/inputPlugins/aac_plugin.c
View File

@ -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;

8
src/inputPlugins/audiofile_plugin.c
View File

@ -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);

4
src/inputPlugins/flac_plugin.c
View File

@ -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

4
src/inputPlugins/mod_plugin.c
View File

@ -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);

2
src/inputPlugins/mp3_plugin.c
View File

@ -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);
}

8
src/inputPlugins/mp4_plugin.c
View File

@ -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);

12
src/inputPlugins/mpc_plugin.c
View File

@ -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,

4
src/inputPlugins/oggflac_plugin.c
View File

@ -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

4
src/inputPlugins/oggvorbis_plugin.c
View File

@ -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)

9
src/inputPlugins/wavpack_plugin.c
View File

@ -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);

10
src/pcm_utils.c
View File

@ -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;

2
src/player_thread.c
View File

@ -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);

2
src/timer.c
View File

@ -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;
}