diff options
Diffstat (limited to 'src/raudio.c')
| -rw-r--r-- | src/raudio.c | 304 |
1 files changed, 128 insertions, 176 deletions
diff --git a/src/raudio.c b/src/raudio.c index c63c5e9d..bb4638dc 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -236,16 +236,16 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr // AudioBuffer management functions declaration // NOTE: Those functions are not exposed by raylib... for the moment AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage); -void CloseAudioBuffer(AudioBuffer *audioBuffer); -bool IsAudioBufferPlaying(AudioBuffer *audioBuffer); -void PlayAudioBuffer(AudioBuffer *audioBuffer); -void StopAudioBuffer(AudioBuffer *audioBuffer); -void PauseAudioBuffer(AudioBuffer *audioBuffer); -void ResumeAudioBuffer(AudioBuffer *audioBuffer); -void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume); -void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch); -void TrackAudioBuffer(AudioBuffer *audioBuffer); -void UntrackAudioBuffer(AudioBuffer *audioBuffer); +void CloseAudioBuffer(AudioBuffer *buffer); +bool IsAudioBufferPlaying(AudioBuffer *buffer); +void PlayAudioBuffer(AudioBuffer *buffer); +void StopAudioBuffer(AudioBuffer *buffer); +void PauseAudioBuffer(AudioBuffer *buffer); +void ResumeAudioBuffer(AudioBuffer *buffer); +void SetAudioBufferVolume(AudioBuffer *buffer, float volume); +void SetAudioBufferPitch(AudioBuffer *buffer, float pitch); +void TrackAudioBuffer(AudioBuffer *buffer); +void UntrackAudioBuffer(AudioBuffer *buffer); //---------------------------------------------------------------------------------- // Multi channel playback globals @@ -644,151 +644,127 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam } // Delete an audio buffer -void CloseAudioBuffer(AudioBuffer *audioBuffer) +void CloseAudioBuffer(AudioBuffer *buffer) { - if (audioBuffer == NULL) + if (buffer != NULL) { - TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer"); - return; + UntrackAudioBuffer(buffer); + RL_FREE(buffer->buffer); + RL_FREE(buffer); } - - UntrackAudioBuffer(audioBuffer); - RL_FREE(audioBuffer->buffer); - RL_FREE(audioBuffer); + else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer"); } // Check if an audio buffer is playing -bool IsAudioBufferPlaying(AudioBuffer *audioBuffer) +bool IsAudioBufferPlaying(AudioBuffer *buffer) { - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer"); - return false; - } + bool result = false; + + if (buffer != NULL) result = (buffer->playing && !buffer->paused); + else TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer"); - return audioBuffer->playing && !audioBuffer->paused; + return result; } // Play an audio buffer // NOTE: Buffer is restarted to the start. // Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained. -void PlayAudioBuffer(AudioBuffer *audioBuffer) +void PlayAudioBuffer(AudioBuffer *buffer) { - if (audioBuffer == NULL) + if (buffer != NULL) { - TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); - return; + buffer->playing = true; + buffer->paused = false; + buffer->frameCursorPos = 0; } - - audioBuffer->playing = true; - audioBuffer->paused = false; - audioBuffer->frameCursorPos = 0; + else TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer"); } // Stop an audio buffer -void StopAudioBuffer(AudioBuffer *audioBuffer) +void StopAudioBuffer(AudioBuffer *buffer) { - if (audioBuffer == NULL) + if (buffer != NULL) { - TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer"); - return; + if (IsAudioBufferPlaying(buffer)) + { + buffer->playing = false; + buffer->paused = false; + buffer->frameCursorPos = 0; + buffer->isSubBufferProcessed[0] = true; + buffer->isSubBufferProcessed[1] = true; + } } - - // Don't do anything if the audio buffer is already stopped. - if (!IsAudioBufferPlaying(audioBuffer)) return; - - audioBuffer->playing = false; - audioBuffer->paused = false; - audioBuffer->frameCursorPos = 0; - audioBuffer->isSubBufferProcessed[0] = true; - audioBuffer->isSubBufferProcessed[1] = true; + else TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer"); } // Pause an audio buffer -void PauseAudioBuffer(AudioBuffer *audioBuffer) +void PauseAudioBuffer(AudioBuffer *buffer) { - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer"); - return; - } - - audioBuffer->paused = true; + if (buffer != NULL) buffer->paused = true; + else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer"); } // Resume an audio buffer -void ResumeAudioBuffer(AudioBuffer *audioBuffer) +void ResumeAudioBuffer(AudioBuffer *buffer) { - if (audioBuffer == NULL) - { - TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer"); - return; - } - - audioBuffer->paused = false; + if (buffer != NULL) buffer->paused = false; + else TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer"); } // Set volume for an audio buffer -void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume) +void SetAudioBufferVolume(AudioBuffer *buffer, float volume) { - if (audioBuffer == NULL) - { - TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer"); - return; - } - - audioBuffer->volume = volume; + if (buffer != NULL) buffer->volume = volume; + else TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer"); } // Set pitch for an audio buffer -void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch) +void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) { - if (audioBuffer == NULL) + if (buffer != NULL) { - TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer"); - return; - } - - float pitchMul = pitch/audioBuffer->pitch; + float pitchMul = pitch/buffer->pitch; - // Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches - // will make the sound faster; lower pitches make it slower. - ma_uint32 newOutputSampleRate = (ma_uint32)((float)audioBuffer->dsp.src.config.sampleRateOut / pitchMul); - audioBuffer->pitch *= (float)audioBuffer->dsp.src.config.sampleRateOut / newOutputSampleRate; + // Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches + // will make the sound faster; lower pitches make it slower. + ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul); + buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate; - ma_pcm_converter_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate); + ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate); + } + else TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer"); } // Track audio buffer to linked list next position -void TrackAudioBuffer(AudioBuffer *audioBuffer) +void TrackAudioBuffer(AudioBuffer *buffer) { ma_mutex_lock(&audioLock); { - if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer; + if (firstAudioBuffer == NULL) firstAudioBuffer = buffer; else { - lastAudioBuffer->next = audioBuffer; - audioBuffer->prev = lastAudioBuffer; + lastAudioBuffer->next = buffer; + buffer->prev = lastAudioBuffer; } - lastAudioBuffer = audioBuffer; + lastAudioBuffer = buffer; } ma_mutex_unlock(&audioLock); } // Untrack audio buffer from linked list -void UntrackAudioBuffer(AudioBuffer *audioBuffer) +void UntrackAudioBuffer(AudioBuffer *buffer) { ma_mutex_lock(&audioLock); { - if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next; - else audioBuffer->prev->next = audioBuffer->next; + if (buffer->prev == NULL) firstAudioBuffer = buffer->next; + else buffer->prev->next = buffer->next; - if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev; - else audioBuffer->next->prev = audioBuffer->prev; + if (buffer->next == NULL) lastAudioBuffer = buffer->prev; + else buffer->next->prev = buffer->prev; - audioBuffer->prev = NULL; - audioBuffer->next = NULL; + buffer->prev = NULL; + buffer->next = NULL; } ma_mutex_unlock(&audioLock); } @@ -802,10 +778,9 @@ Wave LoadWave(const char *fileName) { Wave wave = { 0 }; + if (false) { } #if defined(SUPPORT_FILEFORMAT_WAV) - if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName); -#else - if (false) {} + else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName); #endif #if defined(SUPPORT_FILEFORMAT_OGG) else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName); @@ -821,25 +796,6 @@ Wave LoadWave(const char *fileName) return wave; } -// Load wave data from raw array data -Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels) -{ - Wave wave; - - wave.data = data; - wave.sampleCount = sampleCount; - wave.sampleRate = sampleRate; - wave.sampleSize = sampleSize; - wave.channels = channels; - - // NOTE: Copy wave data to work with, user is responsible of input data to free - Wave cwave = WaveCopy(wave); - - WaveFormat(&cwave, sampleRate, sampleSize, channels); - - return cwave; -} - // Load sound from file // NOTE: The entire file is loaded to memory to be played (no-streaming) Sound LoadSound(const char *fileName) @@ -903,7 +859,7 @@ void UnloadWave(Wave wave) // Unload sound void UnloadSound(Sound sound) { - CloseAudioBuffer((AudioBuffer *)sound.stream.buffer); + CloseAudioBuffer(sound.stream.buffer); TraceLog(LOG_INFO, "Unloaded sound data from RAM"); } @@ -911,7 +867,7 @@ void UnloadSound(Sound sound) // Update sound buffer with new data void UpdateSound(Sound sound, const void *data, int samplesCount) { - AudioBuffer *audioBuffer = (AudioBuffer *)sound.stream.buffer; + AudioBuffer *audioBuffer = sound.stream.buffer; if (audioBuffer == NULL) { @@ -930,10 +886,9 @@ void ExportWave(Wave wave, const char *fileName) { bool success = false; + if (false) { } #if defined(SUPPORT_FILEFORMAT_WAV) - if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName); -#else - if (false) {} + else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName); #endif else if (IsFileExtension(fileName, ".raw")) { @@ -994,7 +949,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName) // Play a sound void PlaySound(Sound sound) { - PlayAudioBuffer((AudioBuffer *)sound.stream.buffer); + PlayAudioBuffer(sound.stream.buffer); } // Play a sound in the multichannel buffer pool @@ -1046,17 +1001,16 @@ void PlaySoundMulti(Sound sound) audioBufferPoolChannels[index] = audioBufferPoolCounter; audioBufferPoolCounter++; - audioBufferPool[index]->volume = ((AudioBuffer*)sound.stream.buffer)->volume; - audioBufferPool[index]->pitch = ((AudioBuffer*)sound.stream.buffer)->pitch; - audioBufferPool[index]->looping = ((AudioBuffer*)sound.stream.buffer)->looping; - audioBufferPool[index]->usage = ((AudioBuffer*)sound.stream.buffer)->usage; + audioBufferPool[index]->volume = sound.stream.buffer->volume; + audioBufferPool[index]->pitch = sound.stream.buffer->pitch; + audioBufferPool[index]->looping = sound.stream.buffer->looping; + audioBufferPool[index]->usage = sound.stream.buffer->usage; audioBufferPool[index]->isSubBufferProcessed[0] = false; audioBufferPool[index]->isSubBufferProcessed[1] = false; - audioBufferPool[index]->bufferSizeInFrames = ((AudioBuffer*)sound.stream.buffer)->bufferSizeInFrames; - audioBufferPool[index]->buffer = ((AudioBuffer*)sound.stream.buffer)->buffer; + audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames; + audioBufferPool[index]->buffer = sound.stream.buffer->buffer; PlayAudioBuffer(audioBufferPool[index]); - } // Stop any sound played with PlaySoundMulti() @@ -1081,37 +1035,37 @@ int GetSoundsPlaying(void) // Pause a sound void PauseSound(Sound sound) { - PauseAudioBuffer((AudioBuffer *)sound.stream.buffer); + PauseAudioBuffer(sound.stream.buffer); } // Resume a paused sound void ResumeSound(Sound sound) { - ResumeAudioBuffer((AudioBuffer *)sound.stream.buffer); + ResumeAudioBuffer(sound.stream.buffer); } // Stop reproducing a sound void StopSound(Sound sound) { - StopAudioBuffer((AudioBuffer *)sound.stream.buffer); + StopAudioBuffer(sound.stream.buffer); } // Check if a sound is playing bool IsSoundPlaying(Sound sound) { - return IsAudioBufferPlaying((AudioBuffer *)sound.stream.buffer); + return IsAudioBufferPlaying(sound.stream.buffer); } // Set volume for a sound void SetSoundVolume(Sound sound, float volume) { - SetAudioBufferVolume((AudioBuffer *)sound.stream.buffer, volume); + SetAudioBufferVolume(sound.stream.buffer, volume); } // Set pitch for a sound void SetSoundPitch(Sound sound, float pitch) { - SetAudioBufferPitch((AudioBuffer *)sound.stream.buffer, pitch); + SetAudioBufferPitch(sound.stream.buffer, pitch); } // Convert wave data to desired format @@ -1213,25 +1167,26 @@ float *GetWaveData(Wave wave) Music LoadMusicStream(const char *fileName) { Music music = (MusicStream *)RL_MALLOC(sizeof(MusicStream)); - bool musicLoaded = true; + bool musicLoaded = false; + if (false) { } #if defined(SUPPORT_FILEFORMAT_OGG) - if (IsFileExtension(fileName, ".ogg")) + else if (IsFileExtension(fileName, ".ogg")) { // Open ogg audio stream music->ctxData = stb_vorbis_open_filename(fileName, NULL, NULL); - if (music->ctxData == NULL) musicLoaded = false; - else + if (music->ctxData != NULL) { + music->ctxType = MUSIC_AUDIO_OGG; stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music->ctxData); // Get Ogg file info // OGG bit rate defaults to 16 bit, it's enough for compressed format music->stream = InitAudioStream(info.sample_rate, 16, info.channels); music->sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music->ctxData)*info.channels; music->sampleLeft = music->sampleCount; - music->ctxType = MUSIC_AUDIO_OGG; - music->loopCount = 0; // Infinite loop by default + music->loopCount = 0; // Infinite loop by default + musicLoaded = true; TraceLog(LOG_DEBUG, "[%s] OGG total samples: %i", fileName, music->sampleCount); TraceLog(LOG_DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate); @@ -1239,24 +1194,22 @@ Music LoadMusicStream(const char *fileName) TraceLog(LOG_DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required); } } -#else - if (false) {} #endif #if defined(SUPPORT_FILEFORMAT_FLAC) else if (IsFileExtension(fileName, ".flac")) { music->ctxData = drflac_open_file(fileName); - if (music->ctxData == NULL) musicLoaded = false; - else + if (music->ctxData != NULL) { + music->ctxType = MUSIC_AUDIO_FLAC; drflac *ctxFlac = (drflac *)music->ctxData; - + music->stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); music->sampleCount = (unsigned int)ctxFlac->totalSampleCount; music->sampleLeft = music->sampleCount; - music->ctxType = MUSIC_AUDIO_FLAC; - music->loopCount = 0; // Infinite loop by default + music->loopCount = 0; // Infinite loop by default + musicLoaded = true; TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->sampleCount); TraceLog(LOG_DEBUG, "[%s] FLAC sample rate: %i", fileName, ctxFlac->sampleRate); @@ -1273,19 +1226,19 @@ Music LoadMusicStream(const char *fileName) int result = drmp3_init_file(ctxMp3, fileName, NULL); - if (!result) musicLoaded = false; - else + if (result > 0) { - TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, ctxMp3->sampleRate); - TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32); - TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, ctxMp3->channels); + music->ctxType = MUSIC_AUDIO_MP3; music->stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); music->sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; music->sampleLeft = music->sampleCount; - music->ctxType = MUSIC_AUDIO_MP3; - music->loopCount = 0; // Infinite loop by default + music->loopCount = 0; // Infinite loop by default + musicLoaded = true; + TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, ctxMp3->sampleRate); + TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32); + TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, ctxMp3->channels); TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music->sampleCount); } } @@ -1297,23 +1250,23 @@ Music LoadMusicStream(const char *fileName) int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName); - if (!result) // XM context created successfully + if (result > 0) // XM context created successfully { - jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops + music->ctxType = MUSIC_MODULE_XM; + jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops // NOTE: Only stereo is supported for XM music->stream = InitAudioStream(48000, 16, 2); music->sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); music->sampleLeft = music->sampleCount; - music->ctxType = MUSIC_MODULE_XM; - music->loopCount = 0; // Infinite loop by default + music->loopCount = 0; // Infinite loop by default + musicLoaded = true; + + music->ctxData = ctxXm; TraceLog(LOG_INFO, "[%s] XM number of samples: %i", fileName, music->sampleCount); TraceLog(LOG_INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music->sampleCount/48000.0f); - - music->ctxData = ctxXm; } - else musicLoaded = false; } #endif #if defined(SUPPORT_FILEFORMAT_MOD) @@ -1323,30 +1276,30 @@ Music LoadMusicStream(const char *fileName) music->ctxData = ctxMod; jar_mod_init(ctxMod); + int result = jar_mod_load_file(ctxMod, fileName); - if (jar_mod_load_file(ctxMod, fileName)) + if (result > 0) { + music->ctxType = MUSIC_MODULE_MOD; + // NOTE: Only stereo is supported for MOD music->stream = InitAudioStream(48000, 16, 2); music->sampleCount = (unsigned int)jar_mod_max_samples(ctxMod); music->sampleLeft = music->sampleCount; - music->ctxType = MUSIC_MODULE_MOD; - music->loopCount = 0; // Infinite loop by default - + music->loopCount = 0; // Infinite loop by default + musicLoaded = true; + TraceLog(LOG_INFO, "[%s] MOD number of samples: %i", fileName, music->sampleLeft); TraceLog(LOG_INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->sampleCount/48000.0f); } - else musicLoaded = false; } #endif - else musicLoaded = false; if (!musicLoaded) { + if (false) { } #if defined(SUPPORT_FILEFORMAT_OGG) - if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData); - #else - if (false) {} + else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music->ctxData); @@ -1377,10 +1330,9 @@ void UnloadMusicStream(Music music) CloseAudioStream(music->stream); + if (false) { } #if defined(SUPPORT_FILEFORMAT_OGG) - if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData); -#else - if (false) {} + else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData); #endif #if defined(SUPPORT_FILEFORMAT_FLAC) else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music->ctxData); @@ -1403,7 +1355,7 @@ void PlayMusicStream(Music music) { if (music != NULL) { - AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.buffer; + AudioBuffer *audioBuffer = music->stream.buffer; if (audioBuffer == NULL) { @@ -1473,7 +1425,7 @@ void UpdateMusicStream(Music music) bool streamEnding = false; - unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.buffer)->bufferSizeInFrames/2; + unsigned int subBufferSizeInFrames = music->stream.buffer->bufferSizeInFrames/2; // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = RL_CALLOC(subBufferSizeInFrames*music->stream.channels*music->stream.sampleSize/8, 1); |