void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int start_time, int end_time) {
const auto max_samples = provider.GetNumSamples();
const auto start_sample = std::min(max_samples, ((int64_t)start_time * provider.GetSampleRate() + 999) / 1000);
const auto end_sample = util::mid(start_sample, ((int64_t)end_time * provider.GetSampleRate() + 999) / 1000, max_samples);
const size_t bytes_per_sample = provider.GetBytesPerSample() * provider.GetChannels();
const size_t bufsize = (end_sample - start_sample) * bytes_per_sample;
writer out{path};
out.write("RIFF");
out.write<int32_t>(bufsize + 36);
out.write("WAVEfmt ");
out.write<int32_t>(16); // Size of chunk
out.write<int16_t>(1); // compression format (PCM)
out.write<int16_t>(provider.GetChannels());
out.write<int32_t>(provider.GetSampleRate());
out.write<int32_t>(provider.GetSampleRate() * provider.GetChannels() * provider.GetBytesPerSample());
out.write<int16_t>(provider.GetChannels() * provider.GetBytesPerSample());
out.write<int16_t>(provider.GetBytesPerSample() * 8);
out.write("data");
out.write<int32_t>(bufsize);
// samples per read
size_t spr = 65536 / bytes_per_sample;
std::vector<char> buf;
for (int64_t i = start_sample; i < end_sample; i += spr) {
spr = std::min<size_t>(spr, end_sample - i);
buf.resize(spr * bytes_per_sample);
provider.GetAudio(&buf[0], i, spr);
out.write(buf);
}
}
/// @brief See if we need to downmix the number of channels
/// @param source_provider
///
AudioProvider *CreateConvertAudioProvider(AudioProvider *source_provider) {
AudioProvider *provider = source_provider;
// Aegisub requires 16 bit samples,
// some audio players break with low samplerates,
// everything breaks with wrong-ended samples.
if (provider->GetBytesPerSample() != 2 ||
provider->GetSampleRate() < 32000 ||
!provider->AreSamplesNativeEndian())
{
// @todo add support for more bitdepths (i.e. 24- and 32-bit audio)
if (provider->GetBytesPerSample() > 2)
throw AudioOpenError("Audio format converter: audio with bitdepths greater than 16 bits/sample is currently unsupported");
provider = new ConvertAudioProvider(provider);
}
// We also require mono audio for historical reasons
if (provider->GetChannels() != 1)
{
provider = new DownmixingAudioProvider(provider);
}
return provider;
}
AudioProvider *AudioProviderFactory::GetProvider(wxString const& filename) {
provider_creator creator;
AudioProvider *provider = nullptr;
provider = creator.try_create("Dummy audio provider", [&]() { return new DummyAudioProvider(filename); });
// Try a PCM provider first
if (!provider && !OPT_GET("Provider/Audio/PCM/Disable")->GetBool())
provider = creator.try_create("PCM audio provider", [&]() { return CreatePCMAudioProvider(filename); });
if (!provider) {
std::vector<std::string> list = GetClasses(OPT_GET("Audio/Provider")->GetString());
if (list.empty()) throw agi::NoAudioProvidersError("No audio providers are available.", 0);
for (auto const& name : list) {
provider = creator.try_create(name, [&]() { return Create(name, filename); });
if (provider) break;
}
}
if (!provider) {
if (creator.found_audio)
throw agi::AudioProviderOpenError(creator.msg, 0);
if (creator.found_file)
throw agi::AudioDataNotFoundError(creator.msg, 0);
throw agi::FileNotFoundError(from_wx(filename));
}
bool needsCache = provider->NeedsCache();
// Give it a converter if needed
if (provider->GetBytesPerSample() != 2 || provider->GetSampleRate() < 32000 || provider->GetChannels() != 1)
provider = CreateConvertAudioProvider(provider);
// Change provider to RAM/HD cache if needed
int cache = OPT_GET("Audio/Cache/Type")->GetInt();
if (!cache || !needsCache)
return new LockAudioProvider(provider);
DialogProgress progress(wxGetApp().frame, _("Load audio"));
// Convert to RAM
if (cache == 1) return new RAMAudioProvider(provider, &progress);
// Convert to HD
if (cache == 2) return new HDAudioProvider(provider, &progress);
throw agi::AudioCacheOpenError("Unknown caching method", 0);
}
/// @brief Open stream
///
void DirectSoundPlayer::OpenStream() {
// Get provider
AudioProvider *provider = GetProvider();
// Initialize the DirectSound object
HRESULT res;
res = DirectSoundCreate8(&DSDEVID_DefaultPlayback,&directSound,NULL); // TODO: support selecting audio device
if (FAILED(res)) throw _T("Failed initializing DirectSound");
// Set DirectSound parameters
AegisubApp *app = (AegisubApp*) wxTheApp;
directSound->SetCooperativeLevel((HWND)app->frame->GetHandle(),DSSCL_PRIORITY);
// Create the wave format structure
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = sizeof(waveFormat);
// Create the buffer initializer
int aim = waveFormat.nAvgBytesPerSec * 15/100; // 150 ms buffer
int min = DSBSIZE_MIN;
int max = DSBSIZE_MAX;
bufSize = std::min(std::max(min,aim),max);
DSBUFFERDESC desc;
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
desc.dwBufferBytes = bufSize;
desc.dwReserved = 0;
desc.lpwfxFormat = &waveFormat;
desc.guid3DAlgorithm = GUID_NULL;
// Create the buffer
IDirectSoundBuffer *buf;
res = directSound->CreateSoundBuffer(&desc,&buf,NULL);
if (res != DS_OK) throw _T("Failed creating DirectSound buffer");
// Copy interface to buffer
res = buf->QueryInterface(IID_IDirectSoundBuffer8,(LPVOID*) &buffer);
if (res != S_OK) throw _T("Failed casting interface to IDirectSoundBuffer8");
// Set data
offset = 0;
}
AudioProvider *AudioProviderFactory::GetProvider(wxString const& filename, int cache) {
AudioProvider *provider = 0;
bool found_file = false;
bool found_audio = false;
std::string msg;
if (!OPT_GET("Provider/Audio/PCM/Disable")->GetBool()) {
// Try a PCM provider first
try {
provider = CreatePCMAudioProvider(filename);
LOG_D("audio_provider") << "Using PCM provider";
}
catch (agi::FileNotFoundError const& err) {
msg = "PCM audio provider: " + err.GetMessage() + " not found.\n";
}
catch (agi::AudioOpenError const& err) {
found_file = true;
msg += err.GetChainedMessage() + "\n";
}
}
if (!provider) {
std::vector<std::string> list = GetClasses(OPT_GET("Audio/Provider")->GetString());
if (list.empty()) throw agi::NoAudioProvidersError("No audio providers are available.", 0);
for (size_t i = 0; i < list.size() ; ++i) {
try {
provider = Create(list[i], filename);
if (provider) {
LOG_D("audio_provider") << "Using audio provider: " << list[i];
break;
}
}
catch (agi::FileNotFoundError const& err) {
msg += list[i] + ": " + err.GetMessage() + " not found.\n";
}
catch (agi::AudioDataNotFoundError const& err) {
found_file = true;
msg += list[i] + ": " + err.GetChainedMessage() + "\n";
}
catch (agi::AudioOpenError const& err) {
found_audio = true;
found_file = true;
msg += list[i] + ": " + err.GetChainedMessage() + "\n";
}
}
}
if (!provider) {
if (found_audio)
throw agi::AudioProviderOpenError(msg, 0);
if (found_file)
throw agi::AudioDataNotFoundError(msg, 0);
throw agi::FileNotFoundError(STD_STR(filename));
}
bool needsCache = provider->NeedsCache();
// Give it a converter if needed
if (provider->GetBytesPerSample() != 2 || provider->GetSampleRate() < 32000 || provider->GetChannels() != 1)
provider = CreateConvertAudioProvider(provider);
// Change provider to RAM/HD cache if needed
if (cache == -1) cache = OPT_GET("Audio/Cache/Type")->GetInt();
if (!cache || !needsCache) {
return new LockAudioProvider(provider);
}
DialogProgress progress(wxGetApp().frame, _("Load audio"));
// Convert to RAM
if (cache == 1) return new RAMAudioProvider(provider, &progress);
// Convert to HD
if (cache == 2) return new HDAudioProvider(provider, &progress);
throw agi::AudioCacheOpenError("Unknown caching method", 0);
}