diff --git a/playback.cpp b/playback.cpp index cdb1f24..26aeb5a 100644 --- a/playback.cpp +++ b/playback.cpp @@ -31,93 +31,47 @@ void PlaybackInstance::SDLCallbackInner(Uint8 *stream, int len) { SDL_memset((void*)stream, 0, len); if (!playback_ready.load()) return; if (process == nullptr) return; + if (st == nullptr) return; size_t unit = sizeof(SAMPLETYPE) * spec.channels; size_t bytes_per_iter = (bufsize / unit) * unit; - SAMPLETYPE *tmpbuf = (SAMPLETYPE*)malloc(bytes_per_iter); - memset(tmpbuf, 0, bytes_per_iter); - while (m_st->numSamples() * unit <= (size_t)len) { - for (auto &data : m_playback_data) { - Uint8 *new_buf; - int new_bufsize; - new_buf = buf; - bytes_per_iter = data.m_backend_spec.size; - if (bytes_per_iter > bufsize) { - bytes_per_iter = bufsize; - } - new_bufsize = bytes_per_iter; - size_t new_bytes = data.m_pProcess->render(buf, bytes_per_iter); - if (SDL_AudioStreamPut(data.m_pStream, buf, new_bytes) < 0) { + while (st->numSamples() * unit <= (size_t)len) { + if (process == nullptr) { + return; + } + Uint8 *new_buf; + int new_bufsize; + new_buf = buf; + bytes_per_iter = backend_spec.size; + if (bytes_per_iter > bufsize) { + bytes_per_iter = bufsize; + } + new_bufsize = bytes_per_iter; + if (this->process != nullptr) { + size_t new_bytes = this->process->render(buf, bytes_per_iter); + if (SDL_AudioStreamPut(sdl_stream, buf, new_bytes) < 0) { ERROR.writefln("SDL_AudioStreamPut: %s", SDL_GetError()); - DEBUG.writefln("Current audio backend: %s", data.m_pProcess->get_backend_id().c_str()); - DEBUG.writefln("SDL_AudioStream *sdl_stream = %0lx", (size_t)data.m_pStream); + DEBUG.writefln("Current audio backend: %s", process->get_backend_id().c_str()); + DEBUG.writefln("SDL_AudioStream *sdl_stream = %0lx", (size_t)sdl_stream); return; } do { - new_bufsize = SDL_AudioStreamGet(data.m_pStream, buf, bytes_per_iter); - if (new_bufsize < 0) { - ERROR.writefln("SDL_AudioStreamGet: %s", SDL_GetError()); - DEBUG.writefln("Current audio backend: %s", data.m_pProcess->get_backend_id().c_str()); - return; - } - SAMPLETYPE *sbuf = (SAMPLETYPE*)buf; - for (size_t i = 0; i < (new_bufsize / unit) * spec.channels; i++) { - if (data.m_fade_time >= 0.0) { - if ((i % spec.channels) == 0) { - data.m_fade_time -= 1.0 / (spec.freq); - if (data.m_fade_time <= 0.0) { - data.delete_needed = true; - break; - } - } - sbuf[i] *= data.m_fade_time; - } - tmpbuf[i] += sbuf[i]; - } - data.m_st->putSamples(sbuf, new_bufsize / unit); - if (data.delete_needed) { - break; - } + new_bufsize = SDL_AudioStreamGet(sdl_stream, buf, std::min((size_t)backend_spec.samples, bufsize / unit) * unit); + if (new_bufsize < 0) { + ERROR.writefln("SDL_AudioStreamGet: %s", SDL_GetError()); + DEBUG.writefln("Current audio backend: %s", process->get_backend_id().c_str()); + return; + } + SAMPLETYPE *sbuf = (SAMPLETYPE*)buf; + for (size_t i = 0; i < (new_bufsize / unit) * spec.channels; i++) { + sbuf[i] *= real_volume; + } + st->putSamples(sbuf, new_bufsize / unit); } while (new_bufsize > 0); + } else { + return; } - for (auto &data : m_playback_data) { - size_t tmplen = 0; - size_t bufused = (bufsize / unit) * unit; - bool done = false; - { - uint samples; - do { - samples = data.m_st->receiveSamples((SAMPLETYPE*)(void*)buf, tmplen / unit); - m_st->putSamples((SAMPLETYPE*)(void*)buf, samples); - } while (samples > 0); - } - for (size_t i = 0; i < len; i += bufused) { - if (i + bufused >= len) { - tmplen = len - i; - } else { - tmplen = bufused; - } - uint samples = m_st->receiveSamples((SAMPLETYPE*)(void*)buf, tmplen / unit); - for (size_t j = 0; j < samples * spec.channels; j++) { - ((SAMPLETYPE*)(void*)stream)[j] += ((SAMPLETYPE*)(void*)buf)[(i * bufused / unit) + j]; - } - if (samples == 0) { - done = true; - break; - } - } - if (data.delete_needed) { - continue; - } - } - for (size_t i = 0; i > m_playback_data.size();) { - auto &data = m_playback_data[i]; - if (data.delete_needed) { - m_playback_data.erase(m_playback_data.begin() + i); - } else { - i++; - } - } - } + } + st->receiveSamples((SAMPLETYPE*)stream, len / unit); } #ifdef __ANDROID__ oboe::DataCallbackResult PlaybackInstance::onAudioReady( @@ -155,23 +109,18 @@ void PlaybackInstance::Load(const char *file, int idx) { LockAudioDevice(); load_finished.store(false); playback_ready.store(false); - for (auto &data : m_playback_data) { - data.m_fade_time = 1.0; - } - PlaybackProcess *process = nullptr; - playback_data data; + if (process != nullptr) delete process; try { process = new PlaybackProcess(file, idx); } catch (std::exception e) { ERROR.writefln("Exception caught when creating process: %s", e.what()); - data.delete_needed = true; + process = nullptr; } length = 0.0; - if (!data.delete_needed && process->process_running()) { + if (process != nullptr && process->process_running()) { length = process->get_length(); auto backend_spec_proxy = process->get_audio_spec(); - SDL_AudioSpec backend_spec; - memset(&backend_spec, 0, sizeof(data.m_backend_spec)); + memset(&backend_spec, 0, sizeof(backend_spec)); backend_spec.channels = backend_spec_proxy->channel_count(); audio_data_t sample_fmt; sample_fmt.size = backend_spec_proxy->bits() / 8; @@ -187,22 +136,23 @@ void PlaybackInstance::Load(const char *file, int idx) { DEBUG.writefln("\tChannels: %d", backend_spec.channels); DEBUG.writefln("\tSample rate: %d", backend_spec.freq); DEBUG.writefln("\tSamples: %d", backend_spec.samples); - try { - data = playback_data(backend_spec, spec, process); - } catch (CustomException &e) { - set_error(e.what()); + if (sdl_stream != nullptr) { + SDL_FreeAudioStream(sdl_stream); + } + sdl_stream = SDL_NewAudioStream(backend_spec.format, backend_spec.channels, backend_spec.freq, spec.format, spec.channels, spec.freq); + if (sdl_stream == nullptr) { + ERROR.writefln("SDL_NewAudioStream: %s", SDL_GetError()); + DEBUG.writefln("format: AUDIO_%s%d%s", sample_fmt.is_float ? "F" : sample_fmt.is_signed ? "S" : "U", sample_fmt.size * 8, sample_fmt.endian ? "MSB" : "LSB"); + set_error("Failed to create SDL audio stream"); set_signal(PlaybackSignalErrorOccurred); - ERROR.writeln(e.what()); - } - size_t required_size = sample_fmt.size * data.m_backend_spec.samples; - m_st->setChannels(spec.channels); - m_st->setRate(spec.freq); - m_st->flush(); - data.m_backend_spec.size = required_size; - if (bufsize < required_size || buf == nullptr) { - bufsize = required_size; - buf = (Uint8*)((buf == nullptr) ? malloc(bufsize) : realloc(buf, bufsize)); } + size_t required_size = sample_fmt.size * backend_spec.samples; + st->setChannels(spec.channels); + st->setRate(spec.freq); + st->flush(); + backend_spec.size = required_size; + bufsize = backend_spec.size; + buf = (Uint8*)malloc(bufsize); if (buf == nullptr) { ERROR.writeln("Failed to allocate memory for playback!"); set_error("Failed to allocate memory for playback!"); @@ -226,8 +176,12 @@ void PlaybackInstance::Load(const char *file, int idx) { flag_mutex.unlock(); } void PlaybackInstance::Unload() { + if (process == nullptr) return; LockAudioDevice(); - m_playback_data.clear(); + delete process; + process = nullptr; + SDL_FreeAudioStream(sdl_stream); + sdl_stream = nullptr; if (buf) free(buf); buf = nullptr; UnlockAudioDevice(); @@ -236,25 +190,25 @@ void PlaybackInstance::UpdateST() { bool any_changed = false; if (speed > 0.0f && speed_changed.exchange(false)) { any_changed = true; - m_st->setRate(speed); + st->setRate(speed); set_signal(PlaybackSignalSpeedChanged); } if (tempo > 0.0f && tempo_changed.exchange(false)) { any_changed = true; - m_st->setTempo(tempo); + st->setTempo(tempo); set_signal(PlaybackSignalTempoChanged); } if (pitch > 0.0f && pitch_changed.exchange(false)) { any_changed = true; - m_st->setPitch(pitch); + st->setPitch(pitch); set_signal(PlaybackSignalPitchChanged); } if (any_changed && process != nullptr) { - process->set_rate(m_st->getInputOutputSampleRatio()); + process->set_rate(st->getInputOutputSampleRatio()); } } double PlaybackInstance::GetMaxSeconds() { - return std::max((double)(MaxSpeed * MaxTempo), m_st->getInputOutputSampleRatio()); + return std::max((double)(MaxSpeed * MaxTempo), st->getInputOutputSampleRatio()); } void PlaybackInstance::InitLoopFunction() { bool reload = false; @@ -278,7 +232,7 @@ void PlaybackInstance::InitLoopFunction() { desired.channels = 2; desired.callback = PlaybackInstance::SDLCallback; desired.userdata = this; - m_st = new SoundTouch(); + st = new SoundTouch(); #ifdef USE_SDL if ((device = SDL_OpenAudioDevice(nullptr, 0, &desired, &obtained, SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)) == 0) { ERROR.writefln("Error opening audio device: '%s'", SDL_GetError()); @@ -335,8 +289,8 @@ Total samples: %u" #error Invalid configuration detected. #endif spec = obtained; - m_st->setSampleRate(spec.freq); - m_st->setChannels(spec.channels); + st->setSampleRate(spec.freq); + st->setChannels(spec.channels); UpdateST(); reload = false; @@ -348,7 +302,9 @@ Total samples: %u" flag_mutex.unlock(); playback_ready.store(false); length = 0.0; - m_playback_data.clear(); + memset(&backend_spec, 0, sizeof(backend_spec)); + if (sdl_stream) SDL_FreeAudioStream(sdl_stream); + sdl_stream = nullptr; bufsize = 0; if (buf) free((void*)buf); buf = nullptr; @@ -450,8 +406,8 @@ void PlaybackInstance::DeinitLoopFunction() { SDL_CloseAudioDevice(device); SDL_QuitSubSystem(SDL_INIT_AUDIO); #endif - delete m_st; - m_st = nullptr; + delete st; + st = nullptr; if (buf) free(buf); current_file_mutex.lock(); current_file = {}; @@ -474,6 +430,7 @@ PlaybackInstance::PlaybackInstance() { running = false; paused = true; position = 0; + sdl_stream = nullptr; length = 0; volume = 100.0; real_volume = 1.0; diff --git a/playback.h b/playback.h index 87bb9f0..317d52b 100644 --- a/playback.h +++ b/playback.h @@ -21,8 +21,6 @@ #include "playback_process.hpp" #include #include -#include "playlist.hpp" -#include "util.hpp" using namespace soundtouch; using std::span; using std::optional; @@ -274,45 +272,12 @@ private: double length; bool paused; Uint8* buf; - size_t tmpbuf_len; size_t bufsize; SDL_AudioDeviceID device; + SoundTouch *st; SDL_AudioSpec spec; - struct playback_data { - SDL_AudioStream *m_pStream = nullptr; - SDL_AudioSpec m_backend_spec = {0}; - PlaybackProcess *m_pProcess = nullptr; - SoundTouch *m_st = nullptr; - bool delete_needed = false; - inline bool fading() { - return m_fade_time >= 0.0; - } - double m_fade_time = -1.0; - inline bool valid() { - return m_st != nullptr && m_pStream != nullptr && m_pProcess != nullptr; - } - inline playback_data() {} - inline playback_data(SDL_AudioSpec backend_spec, SDL_AudioSpec spec, PlaybackProcess *process, double speed = 1.0, double tempo = 1.0, double pitch = 1.0) { - m_backend_spec = backend_spec; - m_pStream = SDL_NewAudioStream(backend_spec.format, backend_spec.channels, backend_spec.freq, spec.format, spec.channels, spec.freq); - if (m_pStream == nullptr) throw CustomException(fmt::format("Could not create stream: {}", SDL_GetError())); - m_st = new SoundTouch(); - m_st->setChannels(spec.channels); - m_st->setSampleRate(spec.freq); - m_st->setRate(speed); - m_st->setPitch(pitch); - m_st->setTempo(tempo); - } - inline ~playback_data() { - if (m_pStream != nullptr) SDL_FreeAudioStream(m_pStream); - delete m_pProcess; - delete m_st; - } - }; - Playlist playlist; - SoundTouch *m_st; - std::vector m_playback_data; - void CleanPlaybackData(); + SDL_AudioSpec backend_spec; + SDL_AudioStream *sdl_stream = nullptr; void SDLCallbackInner(Uint8 *stream, int len); static void SDLCallback(void *userdata, Uint8 *stream, int len); void Load(const char *file, int idx);