diff options
27 files changed, 96 insertions, 97 deletions
diff --git a/src/audio.c b/src/audio.c index bda0107f..a4d4e3ea 100644 --- a/src/audio.c +++ b/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) diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c index 83bd9c25..30ad449f 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 b91895bd..e731f972 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 f26dfcf7..8a2cb6cd 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 59f43a4f..00b069c3 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 487e9a75..8dddf3be 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 9071ed6c..1fc0a5d9 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 38014c8f..93a1d8b3 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 34327573..00c4eb05 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 c54632b1..722079b2 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 14747c32..5983b4d8 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); diff --git a/src/audio_format.h b/src/audio_format.h index dd9b3cae..2475aa77 100644 --- a/src/audio_format.h +++ b/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) diff --git a/src/crossfade.c b/src/crossfade.c index cb780db3..b4d4695c 100644 --- a/src/crossfade.c +++ b/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); diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c index 22d8774a..f24e2053 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 a96623e1..e9b2d747 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 421cdf35..4c08074c 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 cd8a8efd..2f3ec88d 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 9ae9cef1..98bd67f0 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 60e09a1b..2990de1a 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 6a2d167b..d284313d 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 ca37333d..f74dc8dd 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 3a2db5c0..53f233e0 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 0e1d523b..bf244860 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 af7c3a2f..3e99980b 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); diff --git a/src/pcm_utils.c b/src/pcm_utils.c index 9274c2eb..eb3d4b12 100644 --- a/src/pcm_utils.c +++ b/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; diff --git a/src/player_thread.c b/src/player_thread.c index 6a08bf46..efb7d7ab 100644 --- a/src/player_thread.c +++ b/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); diff --git a/src/timer.c b/src/timer.c index f8bbacdc..84c03fbe 100644 --- a/src/timer.c +++ b/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; } |