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Work on the audio implementation.
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@ -1,3 +1,30 @@
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#define MINIAUDIO_IMPLEMENTATION // miniaudio
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#define MA_NO_FLAC // miniaudio
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#define STS_MIXER_IMPLEMENTATION // sts_mixer
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#ifdef __APPLE__
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#define MA_NO_RUNTIME_LINKING // miniaudio osx
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#endif
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//--STRIP
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#include "core/memory.h"
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#include "audio.h"
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#include "3rd_jo_mp1.h"
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#define get_bits stb_vorbis_get_bits
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#define error stb_vorbis_error
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#include "3rd_stb_vorbis.h"
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#undef error
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#undef DEBUG
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#include "3rd_miniaudio.h"
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#include "3rd_sts_mixer.h"
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//--STRIP
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// @fixme: really shutdown audio & related threads before quitting. ma_dr_wav crashes.
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// encapsulate ma_dr_wav,ma_dr_mp3,stbvorbis and some buffer with the sts_mixer_stream_t
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@ -95,15 +122,21 @@ static void reset_stream(mystream_t *stream) {
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// load a (stereo) stream
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static bool load_stream(mystream_t *stream, const char *filename) {
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int datalen;
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char *data = vfs_load(filename, &datalen);
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if (!data)
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int datalen = 0;
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//char *data = vfs_load(filename, &datalen);
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char *data = NULL;
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if (!data) {
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return false;
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}
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int error;
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int HZ = 44100;
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//int HZ = 44100;
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stream->type = UNK;
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stream->loop = true;
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if (stream->type == UNK && (stream->ogg = stb_vorbis_open_memory((const unsigned char *)data, datalen, &error, NULL))) {
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stb_vorbis_info info = stb_vorbis_get_info(stream->ogg);
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if (info.channels != 2) {
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@ -114,7 +147,7 @@ static bool load_stream(mystream_t *stream, const char *filename) {
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stream->stream.sample.frequency = info.sample_rate;
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stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_16;
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}
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if (stream->type == UNK && ma_dr_wav_init_memory(&stream->wav, data, datalen, NULL)) {
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if (stream->type == UNK && ma_dr_wav_init_memory(&stream->wav, data, (size_t)datalen, NULL)) {
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if (stream->wav.channels != 2) {
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puts("cannot stream wav file. stereo required.");
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goto end;
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@ -123,11 +156,15 @@ static bool load_stream(mystream_t *stream, const char *filename) {
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stream->stream.sample.frequency = stream->wav.sampleRate;
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stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_16;
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}
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ma_dr_mp3_config mp3_cfg = { 2, HZ };
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if (stream->type == UNK && (ma_dr_mp3_init_memory(&stream->mp3_, data, datalen, NULL /*&mp3_cfg*/) != 0)) {
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stream->type = MP3;
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stream->stream.sample.frequency = stream->mp3_.sampleRate;
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stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_FLOAT;
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if (stream->type == UNK) {
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//ma_dr_mp3_config mp3_cfg = { 2, HZ };
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if ((ma_dr_mp3_init_memory(&stream->mp3_, data, (size_t)datalen, NULL /*&mp3_cfg*/) != 0)) {
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stream->type = MP3;
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stream->stream.sample.frequency = stream->mp3_.sampleRate;
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stream->stream.sample.audio_format = STS_MIXER_SAMPLE_FORMAT_FLOAT;
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}
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}
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if (stream->type == UNK) {
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@ -147,15 +184,19 @@ end:;
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// load a (mono) sample
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static bool load_sample(sts_mixer_sample_t *sample, const char *filename) {
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int datalen;
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char *data = vfs_load(filename, &datalen);
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if (!data)
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//char *data = vfs_load(filename, &datalen);
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char *data = NULL;
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if (!data) {
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return false;
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}
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int error;
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int channels = 0;
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if (!channels)
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for (ma_dr_wav w = { 0 }, *wav = &w; wav && ma_dr_wav_init_memory(wav, data, datalen, NULL); wav = 0) {
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for (ma_dr_wav w = { 0 }, *wav = &w; wav && ma_dr_wav_init_memory(wav, data, (size_t)datalen, NULL); wav = 0) {
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channels = wav->channels;
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sample->frequency = wav->sampleRate;
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sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
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@ -181,13 +222,14 @@ static bool load_sample(sts_mixer_sample_t *sample, const char *filename) {
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ma_dr_mp3_config mp3_cfg = { 2, 44100 };
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ma_uint64 mp3_fc;
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if (!channels)
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for (short *fbuf = ma_dr_mp3_open_memory_and_read_pcm_frames_s16(data, datalen, &mp3_cfg, &mp3_fc, NULL); fbuf; fbuf = 0) {
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for (short *fbuf = ma_dr_mp3_open_memory_and_read_pcm_frames_s16(data, (size_t)datalen, &mp3_cfg, &mp3_fc, NULL); fbuf; fbuf = 0) {
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channels = mp3_cfg.channels;
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sample->frequency = mp3_cfg.sampleRate;
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sample->audio_format = STS_MIXER_SAMPLE_FORMAT_16;
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sample->length = mp3_fc; // / sizeof(float) / mp3_cfg.channels;
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sample->data = fbuf;
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}
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if (!channels) {
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short *output = 0;
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int outputSize, hz, mp1channels;
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@ -287,8 +329,8 @@ int audio_init(int flags) {
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#endif
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};
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if (ma_context_init(backends, countof(backends), NULL, &context) != MA_SUCCESS) {
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PRINTF("%s\n", "Failed to initialize audio context.");
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if (ma_context_init(backends, (int)(sizeof(backends) / sizeof(0 [backends])), NULL, &context) != MA_SUCCESS) {
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LOG_ERR("Failed to initialize audio context.");
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return false;
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}
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@ -297,7 +339,7 @@ int audio_init(int flags) {
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config.playback.format = ma_format_s32;
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config.playback.channels = 2;
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config.sampleRate = 44100;
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config.dataCallback = (void *)audio_callback; //< @r-lyeh add void* cast
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config.dataCallback = audio_callback;
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config.pUserData = NULL;
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if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) {
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@ -320,20 +362,20 @@ typedef struct audio_handle {
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};
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} audio_handle;
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static array(audio_handle *) audio_instances;
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static Vector<audio_handle *> audio_instances;
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audio_t audio_clip(const char *pathfile) {
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audio_handle *a = REALLOC(0, sizeof(audio_handle));
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audio_handle *a = memnew(audio_handle);
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memset(a, 0, sizeof(audio_handle));
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a->is_clip = load_sample(&a->clip, pathfile);
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array_push(audio_instances, a);
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audio_instances.push_back(a);
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return a;
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}
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audio_t audio_stream(const char *pathfile) {
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audio_handle *a = REALLOC(0, sizeof(audio_handle));
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audio_handle *a = memnew(audio_handle);
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memset(a, 0, sizeof(audio_handle));
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a->is_stream = load_stream(&a->stream, pathfile);
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array_push(audio_instances, a);
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audio_instances.push_back(a);
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return a;
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}
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@ -342,7 +384,7 @@ float audio_volume_clip(float gain) {
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if (gain >= 0 && gain <= 1)
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volume_clip = gain * gain;
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// patch all live clips
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for (int i = 0, active = 0; i < STS_MIXER_VOICES; ++i) {
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for (int i = 0; i < STS_MIXER_VOICES; ++i) {
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if (mixer.voices[i].state != STS_MIXER_VOICE_STOPPED) // is_active?
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if (mixer.voices[i].sample) // is_sample?
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mixer.voices[i].gain = volume_clip;
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@ -353,7 +395,7 @@ float audio_volume_stream(float gain) {
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if (gain >= 0 && gain <= 1)
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volume_stream = gain * gain;
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// patch all live streams
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for (int i = 0, active = 0; i < STS_MIXER_VOICES; ++i) {
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for (int i = 0; i < STS_MIXER_VOICES; ++i) {
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if (mixer.voices[i].state != STS_MIXER_VOICE_STOPPED) // is_active?
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if (mixer.voices[i].stream) // is_stream?
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mixer.voices[i].gain = volume_stream;
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@ -369,9 +411,11 @@ float audio_volume_master(float gain) {
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}
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int audio_mute(int mute) {
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static bool muted = 0;
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do_once muted = flag("--mute") || flag("--muted");
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if (mute >= 0 && mute <= 1)
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if (mute >= 0 && mute <= 1) {
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muted = mute;
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}
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return muted;
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}
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int audio_muted() {
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@ -479,13 +523,14 @@ static bool audio_queue_callback(sts_mixer_sample_t *sample, void *userdata) {
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int sl = sample->length / 2; // 2 ch
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int bytes = sl * 2 * (sample->audio_format == STS_MIXER_SAMPLE_FORMAT_16 ? 2 : 4);
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char *dst = sample->data;
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char *dst = (char *)sample->data;
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static audio_queue_t *aq = 0;
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do {
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while (!aq)
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while (!aq) {
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aq = (audio_queue_t *)thread_queue_consume(&queue_mutex, THREAD_QUEUE_WAIT_INFINITE);
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}
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int len = aq->avail > bytes ? bytes : aq->avail;
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memcpy(dst, (char *)aq->data + aq->cursor, len);
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@ -495,7 +540,7 @@ static bool audio_queue_callback(sts_mixer_sample_t *sample, void *userdata) {
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aq->avail -= len;
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if (aq->avail <= 0) {
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FREE(aq); // @fixme: mattias' original thread_queue_consume() implementation crashes here on tcc+win because of a double free on same pointer. using mcmp for now
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memfree(aq); // @fixme: mattias' original thread_queue_consume() implementation crashes here on tcc+win because of a double free on same pointer. using mcmp for now
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aq = 0;
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}
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} while (bytes > 0);
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@ -542,7 +587,7 @@ int audio_queue(const void *samples, int num_samples, int flags) {
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return 0;
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}
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audio_queue_t *aq = MALLOC(sizeof(audio_queue_t) + (bytes << (channels == 1))); // dupe space if going to be converted from mono to stereo
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audio_queue_t *aq = (audio_queue_t *)memalloc(sizeof(audio_queue_t) + (bytes << (channels == 1))); // dupe space if going to be converted from mono to stereo
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aq->cursor = 0;
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aq->avail = bytes;
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aq->flags = flags;
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