314 lines
7.9 KiB
C
314 lines
7.9 KiB
C
// SPDX-License-Identifier: BSD-2-Clause
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// This code is part of the sfizz library and is licensed under a BSD 2-clause
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// license. You should have receive a LICENSE.md file along with the code.
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// If not, contact the sfizz maintainers at https://github.com/sfztools/sfizz
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#include "st_audiofile.h"
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#if !defined(ST_AUDIO_FILE_USE_SNDFILE)
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#include "st_audiofile_libs.h"
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#include <stdlib.h>
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struct st_audio_file {
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int type;
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union {
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drwav *wav;
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drflac *flac;
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drmp3 *mp3;
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stb_vorbis* ogg;
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};
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union {
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struct { uint64_t frames; } mp3;
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struct { uint32_t channels; float sample_rate; uint64_t frames; } ogg;
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} cache;
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};
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enum {
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st_audio_file_null = -1,
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};
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st_audio_file* st_open_file(const char* filename)
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{
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st_audio_file* af = (st_audio_file*)malloc(sizeof(st_audio_file));
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if (!af)
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return NULL;
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af->type = st_audio_file_null;
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if (af->type == st_audio_file_null) {
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af->wav = (drwav*)malloc(sizeof(drwav));
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if (!af->wav) {
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free(af);
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return NULL;
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}
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if (!drwav_init_file(af->wav, filename, NULL))
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free(af->wav);
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else
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af->type = st_audio_file_wav;
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}
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if (af->type == st_audio_file_null) {
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af->flac = drflac_open_file(filename, NULL);
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if (af->flac)
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af->type = st_audio_file_flac;
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}
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if (af->type == st_audio_file_null) {
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af->ogg = stb_vorbis_open_filename(filename, NULL, NULL);
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if (af->ogg) {
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if ((af->cache.ogg.frames = stb_vorbis_stream_length_in_samples(af->ogg)) == 0)
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stb_vorbis_close(af->ogg);
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else {
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stb_vorbis_info info = stb_vorbis_get_info(af->ogg);
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af->cache.ogg.channels = info.channels;
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af->cache.ogg.sample_rate = info.sample_rate;
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af->type = st_audio_file_ogg;
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}
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}
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}
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if (af->type == st_audio_file_null) {
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af->mp3 = (drmp3*)malloc(sizeof(drmp3));
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if (!af->mp3) {
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free(af);
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return NULL;
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}
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if (!drmp3_init_file(af->mp3, filename, NULL) ||
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(af->cache.mp3.frames = drmp3_get_pcm_frame_count(af->mp3)) == 0)
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free(af->mp3);
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else
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af->type = st_audio_file_mp3;
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}
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if (af->type == st_audio_file_null) {
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free(af);
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af = NULL;
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}
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return af;
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}
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#if defined(_WIN32)
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st_audio_file* st_open_file_w(const wchar_t* filename)
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{
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st_audio_file* af = (st_audio_file*)malloc(sizeof(st_audio_file));
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if (!af)
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return NULL;
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af->type = st_audio_file_null;
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if (af->type == st_audio_file_null) {
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af->wav = (drwav*)malloc(sizeof(drwav));
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if (!af->wav) {
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free(af);
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return NULL;
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}
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if (!drwav_init_file_w(af->wav, filename, NULL))
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free(af->wav);
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else
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af->type = st_audio_file_wav;
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}
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if (af->type == st_audio_file_null) {
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af->flac = drflac_open_file_w(filename, NULL);
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if (af->flac)
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af->type = st_audio_file_flac;
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}
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if (af->type == st_audio_file_null) {
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af->ogg = stb_vorbis_open_filename_w(filename, NULL, NULL);
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if (af->ogg) {
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if ((af->cache.ogg.frames = stb_vorbis_stream_length_in_samples(af->ogg)) == 0)
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stb_vorbis_close(af->ogg);
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else {
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stb_vorbis_info info = stb_vorbis_get_info(af->ogg);
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af->cache.ogg.channels = info.channels;
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af->cache.ogg.sample_rate = info.sample_rate;
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af->type = st_audio_file_ogg;
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}
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}
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}
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if (af->type == st_audio_file_null) {
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af->mp3 = (drmp3*)malloc(sizeof(drmp3));
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if (!af->mp3) {
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free(af);
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return NULL;
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}
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if (!drmp3_init_file_w(af->mp3, filename, NULL) ||
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(af->cache.mp3.frames = drmp3_get_pcm_frame_count(af->mp3)) == 0)
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free(af->mp3);
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else
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af->type = st_audio_file_mp3;
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}
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if (af->type == st_audio_file_null) {
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free(af);
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af = NULL;
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}
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return af;
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}
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#endif
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void st_close(st_audio_file* af)
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{
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switch (af->type) {
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case st_audio_file_wav:
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drwav_uninit(af->wav);
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free(af->wav);
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break;
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case st_audio_file_flac:
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drflac_close(af->flac);
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break;
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case st_audio_file_ogg:
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stb_vorbis_close(af->ogg);
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break;
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case st_audio_file_mp3:
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drmp3_uninit(af->mp3);
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free(af->mp3);
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break;
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}
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af->type = st_audio_file_null;
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}
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int st_get_type(st_audio_file* af)
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{
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return af->type;
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}
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uint32_t st_get_channels(st_audio_file* af)
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{
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uint32_t channels = 0;
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switch (af->type) {
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case st_audio_file_wav:
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channels = af->wav->channels;
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break;
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case st_audio_file_flac:
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channels = af->flac->channels;
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break;
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case st_audio_file_ogg:
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channels = af->cache.ogg.channels;
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break;
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case st_audio_file_mp3:
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channels = af->mp3->channels;
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break;
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}
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return channels;
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}
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float st_get_sample_rate(st_audio_file* af)
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{
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float sample_rate = 0;
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switch (af->type) {
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case st_audio_file_wav:
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sample_rate = af->wav->sampleRate;
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break;
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case st_audio_file_flac:
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sample_rate = af->flac->sampleRate;
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break;
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case st_audio_file_ogg:
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sample_rate = af->cache.ogg.sample_rate;
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break;
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case st_audio_file_mp3:
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sample_rate = af->mp3->sampleRate;
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break;
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}
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return sample_rate;
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}
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uint64_t st_get_frame_count(st_audio_file* af)
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{
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uint64_t frames = 0;
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switch (af->type) {
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case st_audio_file_wav:
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frames = af->wav->totalPCMFrameCount;
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break;
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case st_audio_file_flac:
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frames = af->flac->totalPCMFrameCount;
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break;
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case st_audio_file_ogg:
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frames = af->cache.ogg.frames;
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break;
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case st_audio_file_mp3:
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frames = af->cache.mp3.frames;
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break;
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}
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return frames;
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}
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bool st_seek(st_audio_file* af, uint64_t frame)
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{
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bool success = false;
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switch (af->type) {
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case st_audio_file_wav:
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success = drwav_seek_to_pcm_frame(af->wav, frame);
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break;
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case st_audio_file_flac:
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success = drflac_seek_to_pcm_frame(af->flac, frame);
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break;
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case st_audio_file_ogg:
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success = stb_vorbis_seek(af->ogg, (unsigned)frame) != 0;
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break;
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case st_audio_file_mp3:
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success = drmp3_seek_to_pcm_frame(af->mp3, frame);
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break;
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}
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return success;
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}
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uint64_t st_read_s16(st_audio_file* af, int16_t* buffer, uint64_t count)
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{
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switch (af->type) {
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case st_audio_file_wav:
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count = drwav_read_pcm_frames_s16(af->wav, count, buffer);
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break;
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case st_audio_file_flac:
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count = drflac_read_pcm_frames_s16(af->flac, count, buffer);
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break;
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case st_audio_file_ogg:
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count = stb_vorbis_get_samples_short_interleaved(
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af->ogg, af->cache.ogg.channels, buffer,
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count * af->cache.ogg.channels);
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break;
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case st_audio_file_mp3:
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count = drmp3_read_pcm_frames_s16(af->mp3, count, buffer);
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break;
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}
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return count;
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}
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uint64_t st_read_f32(st_audio_file* af, float* buffer, uint64_t count)
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{
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switch (af->type) {
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case st_audio_file_wav:
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count = drwav_read_pcm_frames_f32(af->wav, count, buffer);
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break;
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case st_audio_file_flac:
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count = drflac_read_pcm_frames_f32(af->flac, count, buffer);
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break;
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case st_audio_file_ogg:
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count = stb_vorbis_get_samples_float_interleaved(
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af->ogg, af->cache.ogg.channels, buffer,
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count * af->cache.ogg.channels);
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break;
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case st_audio_file_mp3:
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count = drmp3_read_pcm_frames_f32(af->mp3, count, buffer);
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break;
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}
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return count;
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}
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#endif // !defined(ST_AUDIO_FILE_USE_SNDFILE)
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