Prepare for playlists in playback

playback.cpp

@@ -31,47 +31,93 @@ 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;
-	while (st->numSamples() * unit <= (size_t)len) {
+    SAMPLETYPE *tmpbuf = (SAMPLETYPE*)malloc(bytes_per_iter);
-		if (process == nullptr) {
+    memset(tmpbuf, 0, bytes_per_iter);
-			return;
+    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 = backend_spec.size;
+            bytes_per_iter = data.m_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 = data.m_pProcess->render(buf, bytes_per_iter);
-            size_t new_bytes = this->process->render(buf, bytes_per_iter);
+            if (SDL_AudioStreamPut(data.m_pStream, buf, new_bytes) < 0) {
-            if (SDL_AudioStreamPut(sdl_stream, buf, new_bytes) < 0) {
                 ERROR.writefln("SDL_AudioStreamPut: %s", SDL_GetError());
-                DEBUG.writefln("Current audio backend: %s", process->get_backend_id().c_str());
+                DEBUG.writefln("Current audio backend: %s", data.m_pProcess->get_backend_id().c_str());
-                DEBUG.writefln("SDL_AudioStream *sdl_stream = %0lx", (size_t)sdl_stream);
+                DEBUG.writefln("SDL_AudioStream *sdl_stream = %0lx", (size_t)data.m_pStream);
                 return;
             }
             do {
-	            new_bufsize = SDL_AudioStreamGet(sdl_stream, buf, std::min((size_t)backend_spec.samples, bufsize / unit) * unit);
+                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", process->get_backend_id().c_str());
+                    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++) {
-	                sbuf[i] *= real_volume;
+                    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;
                 }
-                st->putSamples(sbuf, new_bufsize / unit);
             } while (new_bufsize > 0);
+        }
+        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 {
-        	return;
+                    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(
@@ -109,18 +155,23 @@ void PlaybackInstance::Load(const char *file, int idx) {
     LockAudioDevice();
     load_finished.store(false);
     playback_ready.store(false);
-    if (process != nullptr) delete process;
+    for (auto &data : m_playback_data) {
+        data.m_fade_time = 1.0;
+    }
+    PlaybackProcess *process = nullptr;
+    playback_data data;
     try {
     	process = new PlaybackProcess(file, idx);
     } catch (std::exception e) {
     	ERROR.writefln("Exception caught when creating process: %s", e.what());
-     	process = nullptr;
+     	data.delete_needed = true;
     }
     length = 0.0;
-    if (process != nullptr && process->process_running()) {
+    if (!data.delete_needed && process->process_running()) {
         length = process->get_length();
         auto backend_spec_proxy = process->get_audio_spec();
-        memset(&backend_spec, 0, sizeof(backend_spec));
+        SDL_AudioSpec backend_spec;
+        memset(&backend_spec, 0, sizeof(data.m_backend_spec));
         backend_spec.channels = backend_spec_proxy->channel_count();
         audio_data_t sample_fmt;
         sample_fmt.size = backend_spec_proxy->bits() / 8;
@@ -136,23 +187,22 @@ 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);
-        if (sdl_stream != nullptr) {
+        try {
-            SDL_FreeAudioStream(sdl_stream);
+            data = playback_data(backend_spec, spec, process);
-        }
+        } catch (CustomException &e) {
-        sdl_stream = SDL_NewAudioStream(backend_spec.format, backend_spec.channels, backend_spec.freq, spec.format, spec.channels, spec.freq);
+            set_error(e.what());
-        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!");
@@ -176,12 +226,8 @@ void PlaybackInstance::Load(const char *file, int idx) {
     flag_mutex.unlock();
 }
 void PlaybackInstance::Unload() {
-    if (process == nullptr) return;
     LockAudioDevice();
-    delete process;
+    m_playback_data.clear();
-    process = nullptr;
-    SDL_FreeAudioStream(sdl_stream);
-    sdl_stream = nullptr;
     if (buf) free(buf);
     buf = nullptr;
     UnlockAudioDevice();
@@ -190,25 +236,25 @@ void PlaybackInstance::UpdateST() {
     bool any_changed = false;
     if (speed > 0.0f && speed_changed.exchange(false)) {
         any_changed = true;
-        st->setRate(speed);
+        m_st->setRate(speed);
         set_signal(PlaybackSignalSpeedChanged);
     }
     if (tempo > 0.0f && tempo_changed.exchange(false)) {
         any_changed = true;
-        st->setTempo(tempo);
+        m_st->setTempo(tempo);
         set_signal(PlaybackSignalTempoChanged);
     }
     if (pitch > 0.0f && pitch_changed.exchange(false)) {
         any_changed = true;
-        st->setPitch(pitch);
+        m_st->setPitch(pitch);
         set_signal(PlaybackSignalPitchChanged);
     }
     if (any_changed && process != nullptr) {
-    	process->set_rate(st->getInputOutputSampleRatio());
+    	process->set_rate(m_st->getInputOutputSampleRatio());
     }
 }
 double PlaybackInstance::GetMaxSeconds() {
-    return std::max((double)(MaxSpeed * MaxTempo), st->getInputOutputSampleRatio());
+    return std::max((double)(MaxSpeed * MaxTempo), m_st->getInputOutputSampleRatio());
 }
 void PlaybackInstance::InitLoopFunction() {
     bool reload = false;
@@ -232,7 +278,7 @@ void PlaybackInstance::InitLoopFunction() {
     desired.channels = 2;
     desired.callback = PlaybackInstance::SDLCallback;
     desired.userdata = this;
-    st = new SoundTouch();
+    m_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());
@@ -289,8 +335,8 @@ Total samples: %u"
 #error Invalid configuration detected.
 #endif
     spec = obtained;
-    st->setSampleRate(spec.freq);
+    m_st->setSampleRate(spec.freq);
-    st->setChannels(spec.channels);
+    m_st->setChannels(spec.channels);
     UpdateST();
 
     reload = false;
@@ -302,9 +348,7 @@ Total samples: %u"
     flag_mutex.unlock();
     playback_ready.store(false);
     length = 0.0;
-    memset(&backend_spec, 0, sizeof(backend_spec));
+    m_playback_data.clear();
-    if (sdl_stream) SDL_FreeAudioStream(sdl_stream);
-    sdl_stream = nullptr;
     bufsize = 0;
     if (buf) free((void*)buf);
     buf = nullptr;
@@ -406,8 +450,8 @@ void PlaybackInstance::DeinitLoopFunction() {
 	SDL_CloseAudioDevice(device);
 	SDL_QuitSubSystem(SDL_INIT_AUDIO);
 #endif
-    delete st;
+    delete m_st;
-    st = nullptr;
+    m_st = nullptr;
     if (buf) free(buf);
     current_file_mutex.lock();
     current_file = {};
@@ -430,7 +474,6 @@ PlaybackInstance::PlaybackInstance() {
     running = false;
     paused = true;
     position = 0;
-    sdl_stream = nullptr;
     length = 0;
     volume = 100.0;
     real_volume = 1.0;

playback.h

@@ -21,6 +21,8 @@
 #include "playback_process.hpp"
 #include <google/protobuf/any.h>
 #include <ipc/internal.pb.h>
+#include "playlist.hpp"
+#include "util.hpp"
 using namespace soundtouch;
 using std::span;
 using std::optional;
@@ -272,12 +274,45 @@ private:
     double length;
     bool paused;
     Uint8* buf;
+    size_t tmpbuf_len;
     size_t bufsize;
     SDL_AudioDeviceID device;
-    SoundTouch *st;
     SDL_AudioSpec spec;
-    SDL_AudioSpec backend_spec;
+    struct playback_data {
-    SDL_AudioStream *sdl_stream = nullptr;
+        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<playback_data> m_playback_data;
+    void CleanPlaybackData();
     void SDLCallbackInner(Uint8 *stream, int len);
     static void SDLCallback(void *userdata, Uint8 *stream, int len);
     void Load(const char *file, int idx);