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Working on MP3 loading

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Loading full mp3 file works but loading for music streaming does not work, it seems total number of samples is not obtained correctly from mp3 file...
This commit is contained in:
Ray 2018-10-17 19:39:16 +02:00
parent 8a88e65a94
commit 8bac72f4fa
1 changed files with 22 additions and 18 deletions
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40
src/audio.c
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@ -1463,21 +1463,25 @@ Music LoadMusicStream(const char *fileName)
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3"))
{
drmp3_init_file(&music->ctxMp3, fileName, NULL);
int result = drmp3_init_file(&music->ctxMp3, fileName, NULL);
if (music->ctxMp3.framesRemaining <= 0) musicLoaded = false;
if (!result) musicLoaded = false;
else
{
music->stream = InitAudioStream(music->ctxMp3.sampleRate, 16, music->ctxMp3.channels);
TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate);
TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32);
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels);
TraceLog(LOG_INFO, "[%s] MP3 frames remaining: %i", fileName, (unsigned int)music->ctxMp3.framesRemaining);
music->stream = InitAudioStream(music->ctxMp3.sampleRate, 32, music->ctxMp3.channels);
// TODO: It seems the total number of samples is not obtained correctly...
music->totalSamples = (unsigned int)music->ctxMp3.framesRemaining*music->ctxMp3.channels;
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_AUDIO_MP3;
music->loopCount = -1; // Infinite loop by default
TraceLog(LOG_DEBUG, "[%s] MP3 total samples: %i", fileName, music->totalSamples);
TraceLog(LOG_DEBUG, "[%s] MP3 sample rate: %i", fileName, music->ctxMp3.sampleRate);
//TraceLog(LOG_DEBUG, "[%s] MP3 bits per sample: %i", fileName, music->ctxMp3.bitsPerSample);
TraceLog(LOG_DEBUG, "[%s] MP3 channels: %i", fileName, music->ctxMp3.channels);
TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music->totalSamples);
}
}
#endif
@ -1940,7 +1944,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32));
// The size of a streaming buffer must be at least double the size of a period.
unsigned int periodSize = device.bufferSizeInFrames / device.periods;
unsigned int periodSize = device.bufferSizeInFrames/device.periods;
unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
if (subBufferSize < periodSize) subBufferSize = periodSize;
@ -1951,7 +1955,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
return stream;
}
audioBuffer->looping = true; // Always loop for streaming buffers.
audioBuffer->looping = true; // Always loop for streaming buffers.
stream.audioBuffer = audioBuffer;
#else
// Setup OpenAL format
@ -2398,17 +2402,17 @@ static Wave LoadFLAC(const char *fileName)
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const char *fileName)
{
Wave wave;
Wave wave = { 0 };
// Decode an entire MP3 file in one go
uint64_t totalSampleCount;
drmp3_config *config = NULL;
wave.data = drmp3_open_and_decode_file_f32(fileName, config, &totalSampleCount);
uint64_t totalSampleCount = 0;
drmp3_config config = { 0 };
wave.data = drmp3_open_and_decode_file_f32(fileName, &config, &totalSampleCount);
wave.channels = config->outputChannels;
wave.sampleRate = config->outputSampleRate;
wave.sampleCount = (int)totalSampleCount/wave.channels;
wave.sampleSize = 16;
wave.channels = config.outputChannels;
wave.sampleRate = config.outputSampleRate;
wave.sampleCount = (int)totalSampleCount;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels);