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https://github.com/RetroDECK/ES-DE.git
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293 lines
10 KiB
C++
293 lines
10 KiB
C++
// SPDX-License-Identifier: MIT
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//
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// EmulationStation Desktop Edition
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// AudioManager.cpp
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//
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// Low-level audio functions (using SDL2).
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//
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#include "AudioManager.h"
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#include "Log.h"
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#include "Settings.h"
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#include "Sound.h"
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#include <SDL2/SDL.h>
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AudioManager::AudioManager() noexcept
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{
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// Init on construction.
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init();
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}
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AudioManager::~AudioManager()
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{
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// Deinit on destruction.
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deinit();
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}
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AudioManager& AudioManager::getInstance()
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{
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static AudioManager instance;
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return instance;
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}
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void AudioManager::init()
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{
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LOG(LogInfo) << "Setting up AudioManager...";
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if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
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LOG(LogError) << "Error initializing SDL audio!\n" << SDL_GetError();
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return;
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}
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LOG(LogInfo) << "Audio driver: " << SDL_GetCurrentAudioDriver();
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SDL_AudioSpec sRequestedAudioFormat;
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SDL_memset(&sRequestedAudioFormat, 0, sizeof(sRequestedAudioFormat));
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SDL_memset(&sAudioFormat, 0, sizeof(sAudioFormat));
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// Set up format and callback. SDL will negotiate these settings with the audio driver, so
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// if for instance the driver/hardware does not support 32-bit floating point output, 16-bit
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// integer may be selected instead. ES-DE will handle this automatically as there are no
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// hardcoded audio settings elsewhere in the code.
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sRequestedAudioFormat.freq = 44100;
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sRequestedAudioFormat.format = AUDIO_F32;
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sRequestedAudioFormat.channels = 2;
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sRequestedAudioFormat.samples = 1024;
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sRequestedAudioFormat.callback = mixAudio;
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sRequestedAudioFormat.userdata = nullptr;
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for (int i {0}; i < SDL_GetNumAudioDevices(0); ++i) {
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LOG(LogInfo) << "Detected playback device: " << SDL_GetAudioDeviceName(i, 0);
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}
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sAudioDevice = SDL_OpenAudioDevice(0, 0, &sRequestedAudioFormat, &sAudioFormat,
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SDL_AUDIO_ALLOW_ANY_CHANGE);
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if (sAudioDevice == 0) {
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LOG(LogError) << "Unable to open audio device: " << SDL_GetError();
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sHasAudioDevice = false;
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}
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if (sAudioFormat.freq != sRequestedAudioFormat.freq) {
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LOG(LogDebug) << "AudioManager::init(): Requested sample rate "
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<< std::to_string(sRequestedAudioFormat.freq)
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<< " could not be set, obtained " << std::to_string(sAudioFormat.freq);
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}
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if (sAudioFormat.format != sRequestedAudioFormat.format) {
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LOG(LogDebug) << "AudioManager::init(): Requested format "
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<< std::to_string(sRequestedAudioFormat.format)
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<< " could not be set, obtained " << std::to_string(sAudioFormat.format);
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}
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if (sAudioFormat.channels != sRequestedAudioFormat.channels) {
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LOG(LogDebug) << "AudioManager::init(): Requested channel count "
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<< std::to_string(sRequestedAudioFormat.channels)
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<< " could not be set, obtained " << std::to_string(sAudioFormat.channels);
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}
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#if defined(_WIN64) || defined(__APPLE__)
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// Beats me why the buffer size is not divided by the channel count on some operating systems.
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if (sAudioFormat.samples != sRequestedAudioFormat.samples) {
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#else
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if (sAudioFormat.samples != sRequestedAudioFormat.samples / sRequestedAudioFormat.channels) {
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#endif
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LOG(LogDebug) << "AudioManager::init(): Requested sample buffer size "
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<< std::to_string(sRequestedAudioFormat.samples /
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sRequestedAudioFormat.channels)
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<< " could not be set, obtained " << std::to_string(sAudioFormat.samples);
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}
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// Just in case someone changed the es_settings.xml file manually to invalid values.
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if (Settings::getInstance()->getInt("SoundVolumeNavigation") > 100)
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Settings::getInstance()->setInt("SoundVolumeNavigation", 100);
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if (Settings::getInstance()->getInt("SoundVolumeNavigation") < 0)
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Settings::getInstance()->setInt("SoundVolumeNavigation", 0);
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if (Settings::getInstance()->getInt("SoundVolumeVideos") > 100)
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Settings::getInstance()->setInt("SoundVolumeVideos", 100);
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if (Settings::getInstance()->getInt("SoundVolumeVideos") < 0)
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Settings::getInstance()->setInt("SoundVolumeVideos", 0);
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setupAudioStream(sRequestedAudioFormat.freq);
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}
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void AudioManager::deinit()
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{
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SDL_LockAudioDevice(sAudioDevice);
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SDL_FreeAudioStream(sConversionStream);
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SDL_UnlockAudioDevice(sAudioDevice);
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SDL_CloseAudio();
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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sAudioDevice = 0;
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}
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void AudioManager::mixAudio(void* /*unused*/, Uint8* stream, int len)
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{
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// Process navigation sounds.
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bool stillPlaying {false};
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// Initialize the buffer to "silence".
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SDL_memset(stream, 0, len);
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// Iterate through all our samples.
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std::vector<std::shared_ptr<Sound>>::const_iterator soundIt = sSoundVector.cbegin();
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while (soundIt != sSoundVector.cend()) {
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std::shared_ptr<Sound> sound {*soundIt};
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if (sound->isPlaying()) {
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// Calculate rest length of current sample.
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Uint32 restLength {sound->getLength() - sound->getPosition()};
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if (restLength > static_cast<Uint32>(len)) {
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// If stream length is smaller than sample length, clip it.
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restLength = len;
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}
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// Mix sample into stream.
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SDL_MixAudioFormat(
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stream, &(sound->getData()[sound->getPosition()]), sAudioFormat.format, restLength,
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static_cast<int>(Settings::getInstance()->getInt("SoundVolumeNavigation") * 1.28f));
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if (sound->getPosition() + restLength < sound->getLength()) {
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// Sample hasn't ended yet.
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stillPlaying = true;
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}
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// Set new sound position. if this is at or beyond the end of the sample,
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// it will stop automatically.
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sound->setPosition(sound->getPosition() + restLength);
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}
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// Advance to next sound.
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++soundIt;
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}
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// Process video stream audio generated by VideoFFmpegComponent.
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int streamLength {SDL_AudioStreamAvailable(sConversionStream)};
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if (streamLength <= 0) {
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// If nothing is playing, pause the device until there is more audio to output.
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if (!stillPlaying)
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SDL_PauseAudioDevice(sAudioDevice, 1);
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return;
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}
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int chunkLength {0};
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// Cap the chunk length to the buffer size.
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if (streamLength > len)
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chunkLength = len;
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else
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chunkLength = streamLength;
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std::vector<Uint8> converted(chunkLength);
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int processedLength {
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SDL_AudioStreamGet(sConversionStream, static_cast<void*>(&converted.at(0)), chunkLength)};
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if (processedLength < 0) {
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LOG(LogError) << "AudioManager::mixAudio(): Couldn't convert sound chunk:";
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LOG(LogError) << SDL_GetError();
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return;
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}
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// Enable only when needed, as this generates a lot of debug output.
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// LOG(LogDebug) << "AudioManager::mixAudio(): chunkLength "
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// "/ processedLength / streamLength: " << chunkLength << " / " <<
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// " / " << processedLength << " / " << streamLength;
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// This mute flag is used to make sure that the audio buffer already sent to the
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// stream is not played when the video player has been stopped. Otherwise there would
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// be a short time period when the audio would keep playing after the video was stopped
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// and before the stream was cleared in clearStream().
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bool muteStream {sMuteStream};
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if (muteStream) {
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SDL_MixAudioFormat(stream, &converted.at(0), sAudioFormat.format, processedLength, 0);
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}
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else {
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SDL_MixAudioFormat(
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stream, &converted.at(0), sAudioFormat.format, processedLength,
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static_cast<int>(Settings::getInstance()->getInt("SoundVolumeVideos") * 1.28f));
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}
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// If nothing is playing, pause the device until there is more audio to output.
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if (!stillPlaying && SDL_AudioStreamAvailable(sConversionStream) == 0)
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SDL_PauseAudioDevice(sAudioDevice, 1);
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}
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void AudioManager::registerSound(std::shared_ptr<Sound> sound)
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{
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// Add sound to sound vector.
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sSoundVector.push_back(sound);
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}
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void AudioManager::unregisterSound(std::shared_ptr<Sound> sound)
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{
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for (unsigned int i {0}; i < sSoundVector.size(); ++i) {
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if (sSoundVector.at(i) == sound) {
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sSoundVector[i]->stop();
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sSoundVector.erase(sSoundVector.cbegin() + i);
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return;
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}
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}
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}
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void AudioManager::play()
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{
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// Unpause audio, the mixer will figure out if samples need to be played...
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SDL_PauseAudioDevice(sAudioDevice, 0);
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}
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void AudioManager::stop()
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{
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// Stop playing all Sounds.
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for (unsigned int i {0}; i < sSoundVector.size(); ++i) {
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if (sSoundVector.at(i)->isPlaying())
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sSoundVector[i]->stop();
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}
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// Pause audio.
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SDL_PauseAudioDevice(sAudioDevice, 1);
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}
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void AudioManager::setupAudioStream(int sampleRate)
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{
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SDL_AudioStatus audioStatus {SDL_GetAudioDeviceStatus(sAudioDevice)};
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// It's very important to pause the audio device before setting up the stream,
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// or we may get random crashes if attempting to play samples at the same time.
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SDL_PauseAudioDevice(sAudioDevice, 1);
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SDL_FreeAudioStream(sConversionStream);
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// Used for streaming audio from videos.
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sConversionStream = SDL_NewAudioStream(AUDIO_F32, 2, sampleRate, sAudioFormat.format,
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sAudioFormat.channels, sAudioFormat.freq);
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if (sConversionStream == nullptr) {
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LOG(LogError) << "Failed to create audio conversion stream:";
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LOG(LogError) << SDL_GetError();
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}
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// If the device was previously in a playing state, then restore it.
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if (audioStatus == SDL_AUDIO_PLAYING)
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SDL_PauseAudioDevice(sAudioDevice, 0);
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}
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void AudioManager::processStream(const void* samples, unsigned count)
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{
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SDL_LockAudioDevice(sAudioDevice);
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if (SDL_AudioStreamPut(sConversionStream, samples, count * sizeof(Uint8)) == -1) {
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LOG(LogError) << "Failed to put samples in the conversion stream:";
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LOG(LogError) << SDL_GetError();
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SDL_UnlockAudioDevice(sAudioDevice);
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return;
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}
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if (count > 0)
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SDL_PauseAudioDevice(sAudioDevice, 0);
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SDL_UnlockAudioDevice(sAudioDevice);
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}
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void AudioManager::clearStream()
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{
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SDL_LockAudioDevice(sAudioDevice);
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SDL_AudioStreamClear(sConversionStream);
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SDL_UnlockAudioDevice(sAudioDevice);
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}
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