Merge branch 'xa2-ds' into feature

This commit is contained in:
arch1t3cht 2022-08-16 21:04:44 +02:00
commit e7e87f5584
22 changed files with 981 additions and 206 deletions

View file

@ -21,13 +21,107 @@
#include "libaegisub/log.h"
#include "libaegisub/util.h"
namespace agi {
void AudioProvider::GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const {
GetAudio(buf, start, count);
namespace {
template<typename Source>
class ConvertFloatToInt16 {
Source* src;
public:
ConvertFloatToInt16(Source* src) :src(src) {}
int16_t operator[](size_t idx) const {
Source expanded = src[idx] * 32768;
return expanded < -32768 ? -32768 :
expanded > 32767 ? 32767 :
static_cast<int16_t>(expanded);
}
};
// 8 bits per sample is assumed to be unsigned with a bias of 128,
// while everything else is assumed to be signed with zero bias
class ConvertIntToInt16 {
void* src;
int bytes_per_sample;
public:
ConvertIntToInt16(void* src, int bytes_per_sample) :src(src), bytes_per_sample(bytes_per_sample) {}
const int16_t& operator[](size_t idx) const {
return *reinterpret_cast<int16_t*>(reinterpret_cast<char*>(src) + (idx + 1) * bytes_per_sample - sizeof(int16_t));
}
};
class ConvertUInt8ToInt16 {
uint8_t* src;
public:
ConvertUInt8ToInt16(uint8_t* src) :src(src) {}
int16_t operator[](size_t idx) const {
return int16_t(src[idx]-128) << 8;
}
};
template<typename Source>
class DownmixToMono {
Source src;
int channels;
public:
DownmixToMono(Source src, int channels) :src(src), channels(channels) {}
int16_t operator[](size_t idx) const {
int ret = 0;
// Just average the channels together
for (int i = 0; i < channels; ++i)
ret += src[idx * channels + i];
return ret / channels;
}
};
}
namespace agi {
void AudioProvider::FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const {
if (!float_samples && bytes_per_sample == 2 && channels == 1) {
FillBuffer(buf, start, count);
return;
}
void* buff = malloc(bytes_per_sample * count * channels);
FillBuffer(buff, start, count);
if (channels == 1) {
if (float_samples) {
if (bytes_per_sample == sizeof(float))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff))[i];
else if (bytes_per_sample == sizeof(double))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff))[i];
}
else {
if (bytes_per_sample == sizeof(uint8_t))
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff))[i];
else
for (int64_t i = 0; i < count; ++i)
buf[i] = ConvertIntToInt16(buff, bytes_per_sample)[i];
}
}
else {
if (float_samples) {
if (bytes_per_sample == sizeof(float))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertFloatToInt16<float> >(ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff)), channels)[i];
else if (bytes_per_sample == sizeof(double))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertFloatToInt16<double> >(ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff)), channels)[i];
}
else {
if (bytes_per_sample == sizeof(uint8_t))
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertUInt8ToInt16>(ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff)), channels)[i];
else
for (int64_t i = 0; i < count; ++i)
buf[i] = DownmixToMono<ConvertIntToInt16>(ConvertIntToInt16(buff, bytes_per_sample), channels)[i];
}
}
free(buff);
}
void AudioProvider::GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const {
GetInt16MonoAudio(buf, start, count);
if (volume == 1.0) return;
if (bytes_per_sample != 2)
throw agi::InternalError("GetAudioWithVolume called on unconverted audio stream");
auto buffer = static_cast<int16_t *>(buf);
for (size_t i = 0; i < (size_t)count; ++i)
@ -75,6 +169,39 @@ void AudioProvider::GetAudio(void *buf, int64_t start, int64_t count) const {
}
}
void AudioProvider::GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const {
if (start < 0) {
memset(buf, 0, sizeof(int16_t) * std::min(-start, count));
buf -= start;
count += start;
start = 0;
}
if (start + count > num_samples) {
int64_t zero_count = std::min(count, start + count - num_samples);
count -= zero_count;
memset(buf + count, 0, sizeof(int16_t) * zero_count);
}
if (count <= 0) return;
try {
FillBufferInt16Mono(buf, start, count);
}
catch (AudioDecodeError const& e) {
// We don't have any good way to report errors here, so just log the
// failure and return silence
LOG_E("audio_provider") << e.GetMessage();
memset(buf, 0, sizeof(int16_t) * count);
return;
}
catch (...) {
LOG_E("audio_provider") << "Unknown audio decoding error";
memset(buf, 0, sizeof(int16_t) * count);
return;
}
}
namespace {
class writer {
io::Save outfile;
@ -114,7 +241,7 @@ void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int star
out.write("WAVEfmt ");
out.write<int32_t>(16); // Size of chunk
out.write<int16_t>(1); // compression format (PCM)
out.write<int16_t>(provider.AreSamplesFloat() ? 3 : 1); // compression format (1: WAVE_FORMAT_PCM, 3: WAVE_FORMAT_IEEE_FLOAT)
out.write<int16_t>(provider.GetChannels());
out.write<int32_t>(provider.GetSampleRate());
out.write<int32_t>(provider.GetSampleRate() * provider.GetChannels() * provider.GetBytesPerSample());
@ -134,4 +261,4 @@ void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int star
out.write(buf);
}
}
}
}

View file

@ -22,119 +22,19 @@
#include <limits>
using namespace agi;
/// Anything integral -> 16 bit signed machine-endian audio converter
namespace {
template<class Target>
class BitdepthConvertAudioProvider final : public AudioProviderWrapper {
int src_bytes_per_sample;
mutable std::vector<uint8_t> src_buf;
class ConvertAudioProvider final : public AudioProviderWrapper {
public:
BitdepthConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
if (bytes_per_sample > 8)
throw AudioProviderError("Audio format converter: audio with bitdepths greater than 64 bits/sample is currently unsupported");
src_bytes_per_sample = bytes_per_sample;
bytes_per_sample = sizeof(Target);
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count == count64);
src_buf.resize(count * src_bytes_per_sample * channels);
source->GetAudio(src_buf.data(), start, count);
auto dest = static_cast<int16_t*>(buf);
for (int64_t i = 0; i < count * channels; ++i) {
int64_t sample = 0;
// 8 bits per sample is assumed to be unsigned with a bias of 127,
// while everything else is assumed to be signed with zero bias
if (src_bytes_per_sample == 1)
sample = src_buf[i] - 128;
else {
for (int j = src_bytes_per_sample; j > 0; --j) {
sample <<= 8;
sample += src_buf[i * src_bytes_per_sample + j - 1];
}
}
if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
sample /= 1LL << (src_bytes_per_sample - sizeof(Target)) * 8;
else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
sample *= 1LL << (sizeof(Target) - src_bytes_per_sample ) * 8;
dest[i] = static_cast<Target>(sample);
}
}
};
/// Floating point -> 16 bit signed machine-endian audio converter
template<class Source, class Target>
class FloatConvertAudioProvider final : public AudioProviderWrapper {
mutable std::vector<Source> src_buf;
public:
FloatConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
bytes_per_sample = sizeof(Target);
ConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
float_samples = false;
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count == count64);
src_buf.resize(count * channels);
source->GetAudio(&src_buf[0], start, count);
auto dest = static_cast<Target*>(buf);
for (size_t i = 0; i < static_cast<size_t>(count * channels); ++i) {
Source expanded;
if (src_buf[i] < 0)
expanded = static_cast<Target>(-src_buf[i] * std::numeric_limits<Target>::min());
else
expanded = static_cast<Target>(src_buf[i] * std::numeric_limits<Target>::max());
dest[i] = expanded < std::numeric_limits<Target>::min() ? std::numeric_limits<Target>::min() :
expanded > std::numeric_limits<Target>::max() ? std::numeric_limits<Target>::max() :
static_cast<Target>(expanded);
}
}
};
/// Non-mono 16-bit signed machine-endian -> mono 16-bit signed machine endian converter
class DownmixAudioProvider final : public AudioProviderWrapper {
int src_channels;
mutable std::vector<int16_t> src_buf;
public:
DownmixAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
src_channels = channels;
channels = 1;
bytes_per_sample = sizeof(int16_t);
}
void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
auto count = static_cast<size_t>(count64);
assert(count == count64);
src_buf.resize(count * src_channels);
source->GetAudio(&src_buf[0], start, count);
auto dst = static_cast<int16_t*>(buf);
// Just average the channels together
while (count-- > 0) {
int sum = 0;
for (int c = 0; c < src_channels; ++c)
sum += src_buf[count * src_channels + c];
dst[count] = static_cast<int16_t>(sum / src_channels);
}
void FillBuffer(void *buf, int64_t start, int64_t count) const override {
source->GetInt16MonoAudio(reinterpret_cast<int16_t*>(buf), start, count);
}
};
/// Sample doubler with linear interpolation for the samples provider
/// Requires 16-bit mono input
class SampleDoublingAudioProvider final : public AudioProviderWrapper {
@ -177,26 +77,23 @@ std::unique_ptr<AudioProvider> CreateConvertAudioProvider(std::unique_ptr<AudioP
// Ensure 16-bit audio with proper endianness
if (provider->AreSamplesFloat()) {
LOG_D("audio_provider") << "Converting float to S16";
if (provider->GetBytesPerSample() == sizeof(float))
provider = agi::make_unique<FloatConvertAudioProvider<float, int16_t>>(std::move(provider));
else
provider = agi::make_unique<FloatConvertAudioProvider<double, int16_t>>(std::move(provider));
}
if (provider->GetBytesPerSample() != 2) {
LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample or wrong endian to S16";
provider = agi::make_unique<BitdepthConvertAudioProvider<int16_t>>(std::move(provider));
LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample to S16";
}
// We currently only support mono audio
if (provider->GetChannels() != 1) {
LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
provider = agi::make_unique<DownmixAudioProvider>(std::move(provider));
}
// Some players don't like low sample rate audio
while (provider->GetSampleRate() < 32000) {
LOG_D("audio_provider") << "Doubling sample rate";
provider = agi::make_unique<SampleDoublingAudioProvider>(std::move(provider));
if (provider->GetSampleRate() < 32000) {
provider = agi::make_unique<ConvertAudioProvider>(std::move(provider));
while (provider->GetSampleRate() < 32000) {
LOG_D("audio_provider") << "Doubling sample rate";
provider = agi::make_unique<SampleDoublingAudioProvider>(std::move(provider));
}
}
return provider;

View file

@ -43,15 +43,15 @@ class HDAudioProvider final : public AudioProviderWrapper {
}
if (count > 0) {
start *= bytes_per_sample;
count *= bytes_per_sample;
start *= bytes_per_sample * channels;
count *= bytes_per_sample * channels;
memcpy(buf, file.read(start, count), count);
}
}
fs::path CacheFilename(fs::path const& dir) {
// Check free space
if ((uint64_t)num_samples * bytes_per_sample > fs::FreeSpace(dir))
if ((uint64_t)num_samples * bytes_per_sample * channels > fs::FreeSpace(dir))
throw AudioProviderError("Not enough free disk space in " + dir.string() + " to cache the audio");
return format("audio-%lld-%lld", time(nullptr),
@ -61,7 +61,7 @@ class HDAudioProvider final : public AudioProviderWrapper {
public:
HDAudioProvider(std::unique_ptr<AudioProvider> src, agi::fs::path const& dir)
: AudioProviderWrapper(std::move(src))
, file(dir / CacheFilename(dir), num_samples * bytes_per_sample)
, file(dir / CacheFilename(dir), num_samples * bytes_per_sample * channels)
{
decoded_samples = 0;
decoder = std::thread([&] {

View file

@ -29,6 +29,11 @@ class LockAudioProvider final : public agi::AudioProviderWrapper {
source->GetAudio(buf, start, count);
}
void FillBufferInt16Mono(int16_t *buf, int64_t start, int64_t count) const override {
std::unique_lock<std::mutex> lock(mutex);
source->GetInt16MonoAudio(buf, start, count);
}
public:
LockAudioProvider(std::unique_ptr<AudioProvider> src)
: AudioProviderWrapper(std::move(src))

View file

@ -46,14 +46,14 @@ public:
decoded_samples = 0;
try {
blockcache.resize((source->GetNumSamples() * source->GetBytesPerSample() + CacheBlockSize - 1) >> CacheBits);
blockcache.resize((num_samples * bytes_per_sample * channels + CacheBlockSize - 1) >> CacheBits);
}
catch (std::bad_alloc const&) {
throw AudioProviderError("Not enough memory available to cache in RAM");
}
decoder = std::thread([&] {
int64_t readsize = CacheBlockSize / source->GetBytesPerSample();
int64_t readsize = CacheBlockSize / bytes_per_sample / channels;
for (size_t i = 0; i < blockcache.size(); i++) {
if (cancelled) break;
auto actual_read = std::min<int64_t>(readsize, num_samples - i * readsize);
@ -71,20 +71,22 @@ public:
void RAMAudioProvider::FillBuffer(void *buf, int64_t start, int64_t count) const {
auto charbuf = static_cast<char *>(buf);
for (int64_t bytes_remaining = count * bytes_per_sample; bytes_remaining; ) {
for (int64_t bytes_remaining = count * bytes_per_sample * channels; bytes_remaining; ) {
if (start >= decoded_samples) {
memset(charbuf, 0, bytes_remaining);
break;
}
const int i = (start * bytes_per_sample) >> CacheBits;
const int start_offset = (start * bytes_per_sample) & (CacheBlockSize-1);
const int read_size = std::min<int>(bytes_remaining, CacheBlockSize - start_offset);
const int64_t samples_per_block = CacheBlockSize / bytes_per_sample / channels;
const size_t i = start / samples_per_block;
const int start_offset = (start % samples_per_block) * bytes_per_sample * channels;
const int read_size = std::min<int>(bytes_remaining, samples_per_block * bytes_per_sample * channels - start_offset);
memcpy(charbuf, &blockcache[i][start_offset], read_size);
charbuf += read_size;
bytes_remaining -= read_size;
start += read_size / bytes_per_sample;
start += read_size / bytes_per_sample / channels;
}
}
}
@ -93,4 +95,4 @@ namespace agi {
std::unique_ptr<AudioProvider> CreateRAMAudioProvider(std::unique_ptr<AudioProvider> src) {
return agi::make_unique<RAMAudioProvider>(std::move(src));
}
}
}

View file

@ -20,8 +20,8 @@
#include <libaegisub/fs_fwd.h>
#include <atomic>
#include <memory>
#include <vector>
#include <memory>
namespace agi {
class AudioProvider {
@ -37,6 +37,7 @@ protected:
bool float_samples = false;
virtual void FillBuffer(void *buf, int64_t start, int64_t count) const = 0;
virtual void FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const;
void ZeroFill(void *buf, int64_t count) const;
@ -44,7 +45,8 @@ public:
virtual ~AudioProvider() = default;
void GetAudio(void *buf, int64_t start, int64_t count) const;
void GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const;
void GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const;
void GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const;
int64_t GetNumSamples() const { return num_samples; }
int64_t GetDecodedSamples() const { return decoded_samples; }

View file

@ -7,7 +7,7 @@ project('Aegisub', ['c', 'cpp'],
cmake = import('cmake')
if host_machine.system() == 'windows'
add_project_arguments('-DNOMINMAX', '-D_WIN32_WINNT=0x0601', language: 'cpp')
add_project_arguments('-DNOMINMAX', language: 'cpp')
if not get_option('csri').disabled()
add_global_arguments('-DCSRI_NO_EXPORT', language: 'c')
@ -266,20 +266,44 @@ if get_option('vapoursynth').enabled()
dep_avail += 'VapourSynth'
endif
if host_machine.system() == 'windows' and not get_option('directsound').disabled()
dsound_dep = cc.find_library('dsound', required: get_option('directsound'))
winmm_dep = cc.find_library('winmm', required: get_option('directsound'))
ole32_dep = cc.find_library('ole32', required: get_option('directsound'))
have_dsound_h = cc.has_header('dsound.h')
if not have_dsound_h and get_option('directsound').enabled()
error('DirectSound enabled but dsound.h not found')
if host_machine.system() == 'windows'
if not get_option('directsound').disabled()
dsound_dep = cc.find_library('dsound', required: get_option('directsound'))
winmm_dep = cc.find_library('winmm', required: get_option('directsound'))
ole32_dep = cc.find_library('ole32', required: get_option('directsound'))
have_dsound_h = cc.has_header('dsound.h')
if not have_dsound_h and get_option('directsound').enabled()
error('DirectSound enabled but dsound.h not found')
endif
dxguid_dep = cc.find_library('dxguid', required: true)
if dsound_dep.found() and winmm_dep.found() and ole32_dep.found() and dxguid_dep.found() and have_dsound_h
deps += [dsound_dep, winmm_dep, ole32_dep, dxguid_dep]
conf.set('WITH_DIRECTSOUND', 1)
dep_avail += 'DirectSound'
endif
endif
dxguid_dep = cc.find_library('dxguid', required: true)
if dsound_dep.found() and winmm_dep.found() and ole32_dep.found() and dxguid_dep.found() and have_dsound_h
deps += [dsound_dep, winmm_dep, ole32_dep, dxguid_dep]
conf.set('WITH_DIRECTSOUND', 1)
dep_avail += 'DirectSound'
if not get_option('xaudio2').disabled()
have_xaudio_h = cc.has_header('xaudio2.h')
xaudio2_dep = cc.find_library('xaudio2', required: true)
if have_xaudio_h and xaudio2_dep.found()
deps += [xaudio2_dep]
conf.set('WITH_XAUDIO2', 1)
dep_avail += 'XAudio2'
# XAudio2 needs Windows 8 or newer, so we tell meson not to define an older windows or else it can break things.
add_project_arguments('-D_WIN32_WINNT=0x0602', language: 'cpp')
else
# Windows 8 not required if XAudio2 fails to be found. revert for compat.
add_project_arguments('-D_WIN32_WINNT=0x0601', language: 'cpp')
endif
if not have_dsound_h and get_option('xaudio2').enabled()
error('xaudio2 enabled but xaudio2.h not found')
endif
else
# Windows 8 not required if XAudio2 is disabled. revert for compat.
add_project_arguments('-D_WIN32_WINNT=0x0601', language: 'cpp')
endif
endif

View file

@ -3,7 +3,8 @@ option('openal', type: 'feature', description: 'OpenAL audio output')
option('libpulse', type: 'feature', description: 'PulseAudio audio output')
option('portaudio', type: 'feature', description: 'PortAudio audio output')
option('directsound', type: 'feature', description: 'DirectSound audio output')
option('default_audio_output', type: 'combo', choices: ['auto', 'ALSA', 'OpenAL', 'PulseAudio', 'PortAudio', 'DirectSound'], description: 'Default audio output')
option('xaudio2', type: 'feature', description: 'XAudio2 audio output')
option('default_audio_output', type: 'combo', choices: ['auto', 'ALSA', 'OpenAL', 'PulseAudio', 'PortAudio', 'DirectSound', 'XAudio2'], description: 'Default audio output')
option('ffms2', type: 'feature', description: 'FFMS2 video source')
option('avisynth', type: 'feature', description: 'AviSynth video source')

View file

@ -43,6 +43,7 @@
std::unique_ptr<AudioPlayer> CreateAlsaPlayer(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreateDirectSoundPlayer(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreateDirectSound2Player(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreateXAudio2Player(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreateOpenALPlayer(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreatePortAudioPlayer(agi::AudioProvider *providers, wxWindow *window);
std::unique_ptr<AudioPlayer> CreatePulseAudioPlayer(agi::AudioProvider *providers, wxWindow *window);
@ -63,6 +64,9 @@ namespace {
{"DirectSound-old", CreateDirectSoundPlayer, false},
{"DirectSound", CreateDirectSound2Player, false},
#endif
#ifdef WITH_XAUDIO2
{"Xaudio2", CreateXAudio2Player, false},
#endif
#ifdef WITH_OPENAL
{"OpenAL", CreateOpenALPlayer, false},
#endif

View file

@ -127,7 +127,7 @@ void AlsaPlayer::PlaybackThread()
do_setup:
snd_pcm_format_t pcm_format;
switch (provider->GetBytesPerSample())
switch (/*provider->GetBytesPerSample()*/ sizeof(int16_t))
{
case 1:
LOG_D("audio/player/alsa") << "format U8";
@ -143,7 +143,7 @@ do_setup:
if (snd_pcm_set_params(pcm,
pcm_format,
SND_PCM_ACCESS_RW_INTERLEAVED,
provider->GetChannels(),
/*provider->GetChannels()*/ 1,
provider->GetSampleRate(),
1, // allow resample
100*1000 // 100 milliseconds latency
@ -151,7 +151,8 @@ do_setup:
return;
LOG_D("audio/player/alsa") << "set pcm params";
size_t framesize = provider->GetChannels() * provider->GetBytesPerSample();
//size_t framesize = provider->GetChannels() * provider->GetBytesPerSample();
size_t framesize = sizeof(int16_t);
while (true)
{
@ -175,7 +176,7 @@ do_setup:
{
auto avail = std::min(snd_pcm_avail(pcm), (snd_pcm_sframes_t)(end_position-position));
decode_buffer.resize(avail * framesize);
provider->GetAudioWithVolume(decode_buffer.data(), position, avail, volume);
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(decode_buffer.data()), position, avail, volume);
snd_pcm_sframes_t written = 0;
while (written <= 0)
@ -235,7 +236,7 @@ do_setup:
{
decode_buffer.resize(avail * framesize);
provider->GetAudioWithVolume(decode_buffer.data(), position, avail, volume);
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(decode_buffer.data()), position, avail, volume);
snd_pcm_sframes_t written = 0;
while (written <= 0)
{
@ -352,4 +353,4 @@ std::unique_ptr<AudioPlayer> CreateAlsaPlayer(agi::AudioProvider *provider, wxWi
return agi::make_unique<AlsaPlayer>(provider);
}
#endif // WITH_ALSA
#endif // WITH_ALSA

View file

@ -45,6 +45,7 @@
#include <mmsystem.h>
#include <dsound.h>
#include <cguid.h>
namespace {
class DirectSoundPlayer;
@ -111,8 +112,10 @@ DirectSoundPlayer::DirectSoundPlayer(agi::AudioProvider *provider, wxWindow *par
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.nChannels = provider->GetChannels();
waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
//waveFormat.nChannels = provider->GetChannels();
//waveFormat.wBitsPerSample = provider->GetBytesPerSample() * 8;
waveFormat.nChannels = 1;
waveFormat.wBitsPerSample = sizeof(int16_t) * 8;
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = sizeof(waveFormat);
@ -160,7 +163,7 @@ bool DirectSoundPlayer::FillBuffer(bool fill) {
HRESULT res;
void *ptr1, *ptr2;
unsigned long int size1, size2;
int bytesps = provider->GetBytesPerSample();
int bytesps = /*provider->GetBytesPerSample()*/ sizeof(int16_t);
// To write length
int toWrite = 0;
@ -223,8 +226,8 @@ RetryLock:
LOG_D_IF(!count1 && !count2, "audio/player/dsound1") << "DS fill: nothing";
// Get source wave
if (count1) provider->GetAudioWithVolume(ptr1, playPos, count1, volume);
if (count2) provider->GetAudioWithVolume(ptr2, playPos+count1, count2, volume);
if (count1) provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(ptr1), playPos, count1, volume);
if (count2) provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(ptr2), playPos+count1, count2, volume);
playPos += count1+count2;
buffer->Unlock(ptr1,count1*bytesps,ptr2,count2*bytesps);
@ -254,7 +257,7 @@ void DirectSoundPlayer::Play(int64_t start,int64_t count) {
FillBuffer(true);
DWORD play_flag = 0;
if (count*provider->GetBytesPerSample() > bufSize) {
if (count*/*provider->GetBytesPerSample()*/sizeof(int16_t) > bufSize) {
// Start thread
thread = new DirectSoundPlayerThread(this);
thread->Create();
@ -371,4 +374,4 @@ std::unique_ptr<AudioPlayer> CreateDirectSoundPlayer(agi::AudioProvider *provide
return agi::make_unique<DirectSoundPlayer>(provider, parent);
}
#endif // WITH_DIRECTSOUND
#endif // WITH_DIRECTSOUND

View file

@ -317,13 +317,14 @@ void DirectSoundPlayer2Thread::Run()
// Describe the wave format
WAVEFORMATEX waveFormat;
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = provider->GetSampleRate();
waveFormat.cbSize = 0;
waveFormat.wFormatTag = provider->AreSamplesFloat() ? 3 : WAVE_FORMAT_PCM; // Eh fuck it.
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);
//waveFormat.cbSize = sizeof(waveFormat);
// And the buffer itself
int aim = waveFormat.nAvgBytesPerSec * (wanted_latency*buffer_length)/1000;
@ -332,7 +333,7 @@ void DirectSoundPlayer2Thread::Run()
DWORD bufSize = mid(min,aim,max); // size of entire playback buffer
DSBUFFERDESC desc;
desc.dwSize = sizeof(DSBUFFERDESC);
desc.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
desc.dwFlags = DSBCAPS_CTRLVOLUME | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
desc.dwBufferBytes = bufSize;
desc.dwReserved = 0;
desc.lpwfxFormat = &waveFormat;
@ -461,6 +462,15 @@ stop_playback:
goto do_fill_buffer;
case WAIT_OBJECT_0+3:
{
LONG invert_volume = (LONG)((this->volume - 1.0) * 5000.0); // Hrmm weirdly it's half?
// Look, I would have used a min max but it just errored out for me lol.
if (invert_volume > DSBVOLUME_MAX)
invert_volume = DSBVOLUME_MAX;
else if (invert_volume < DSBVOLUME_MIN / 2)
invert_volume = DSBVOLUME_MIN / 2;
bfr->SetVolume(invert_volume);
}
// Change volume
// We aren't thread safe right now, filling the buffers grabs volume directly
// from the field set by the controlling thread, but it shouldn't be a major
@ -608,7 +618,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
buf2sz = 0;
}
provider->GetAudioWithVolume(buf1, input_frame, buf1szf, volume);
provider->GetAudio(buf1, input_frame, buf1szf);
input_frame += buf1szf;
}
@ -621,7 +631,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
buf2sz = buf2szf * bytes_per_frame;
}
provider->GetAudioWithVolume(buf2, input_frame, buf2szf, volume);
provider->GetAudio(buf2, input_frame, buf2szf);
input_frame += buf2szf;
}
@ -932,4 +942,4 @@ std::unique_ptr<AudioPlayer> CreateDirectSound2Player(agi::AudioProvider *provid
return agi::make_unique<DirectSoundPlayer2>(provider, parent);
}
#endif // WITH_DIRECTSOUND
#endif // WITH_DIRECTSOUND

View file

@ -125,7 +125,7 @@ public:
OpenALPlayer::OpenALPlayer(agi::AudioProvider *provider)
: AudioPlayer(provider)
, samplerate(provider->GetSampleRate())
, bpf(provider->GetChannels() * provider->GetBytesPerSample())
, bpf(/*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t))
{
device = alcOpenDevice(nullptr);
if (!device) throw AudioPlayerOpenError("Failed opening default OpenAL device");
@ -241,7 +241,7 @@ void OpenALPlayer::FillBuffers(ALsizei count)
if (fill_len > 0)
// Get fill_len frames of audio
provider->GetAudioWithVolume(&decode_buffer[0], cur_frame, fill_len, volume);
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(decode_buffer.data()), cur_frame, fill_len, volume);
if ((size_t)fill_len * bpf < decode_buffer.size())
// And zerofill the rest
memset(&decode_buffer[fill_len * bpf], 0, decode_buffer.size() - fill_len * bpf);
@ -308,4 +308,4 @@ std::unique_ptr<AudioPlayer> CreateOpenALPlayer(agi::AudioProvider *provider, wx
return agi::make_unique<OpenALPlayer>(provider);
}
#endif // WITH_OPENAL
#endif // WITH_OPENAL

View file

@ -131,7 +131,7 @@ public:
while (!TestDestroy() && parent->cur_frame < parent->end_frame) {
int rsize = std::min(wsize, parent->end_frame - parent->cur_frame);
parent->provider->GetAudioWithVolume(buf, parent->cur_frame,
parent->provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf), parent->cur_frame,
rsize, parent->volume);
int written = ::write(parent->dspdev, buf, rsize * parent->bpf);
parent->cur_frame += written / parent->bpf;
@ -146,7 +146,7 @@ public:
void OSSPlayer::OpenStream()
{
bpf = provider->GetChannels() * provider->GetBytesPerSample();
bpf = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
// Open device
wxString device = to_wx(OPT_GET("Player/Audio/OSS/Device")->GetString());
@ -162,14 +162,14 @@ void OSSPlayer::OpenStream()
#endif
// Set number of channels
int channels = provider->GetChannels();
int channels = /*provider->GetChannels()*/1;
if (ioctl(dspdev, SNDCTL_DSP_CHANNELS, &channels) < 0) {
throw AudioPlayerOpenError("OSS player: setting channels failed");
}
// Set sample format
int sample_format;
switch (provider->GetBytesPerSample()) {
switch (/*provider->GetBytesPerSample()*/sizeof(int16_t)) {
case 1:
sample_format = AFMT_S8;
break;
@ -283,4 +283,4 @@ std::unique_ptr<AudioPlayer> CreateOSSPlayer(agi::AudioProvider *provider, wxWin
return agi::make_unique<OSSPlayer>(provider);
}
#endif // WITH_OSS
#endif // WITH_OSS

View file

@ -140,7 +140,7 @@ void PortAudioPlayer::OpenStream() {
const PaDeviceInfo *device_info = Pa_GetDeviceInfo((*device_ids)[i]);
PaStreamParameters pa_output_p;
pa_output_p.device = (*device_ids)[i];
pa_output_p.channelCount = provider->GetChannels();
pa_output_p.channelCount = /*provider->GetChannels()*/ 1;
pa_output_p.sampleFormat = paInt16;
pa_output_p.suggestedLatency = device_info->defaultLowOutputLatency;
pa_output_p.hostApiSpecificStreamInfo = nullptr;
@ -222,7 +222,7 @@ int PortAudioPlayer::paCallback(const void *inputBuffer, void *outputBuffer,
// Play something
if (lenAvailable > 0) {
player->provider->GetAudioWithVolume(outputBuffer, player->current, lenAvailable, player->GetVolume());
player->provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(outputBuffer), player->current, lenAvailable, player->GetVolume());
// Set play position
player->current += lenAvailable;
@ -283,4 +283,4 @@ std::unique_ptr<AudioPlayer> CreatePortAudioPlayer(agi::AudioProvider *provider,
return agi::make_unique<PortAudioPlayer>(provider);
}
#endif // WITH_PORTAUDIO
#endif // WITH_PORTAUDIO

View file

@ -133,11 +133,11 @@ PulseAudioPlayer::PulseAudioPlayer(agi::AudioProvider *provider) : AudioPlayer(p
}
// Set up stream
bpf = provider->GetChannels() * provider->GetBytesPerSample();
bpf = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
pa_sample_spec ss;
ss.format = PA_SAMPLE_S16LE; // FIXME
ss.rate = provider->GetSampleRate();
ss.channels = provider->GetChannels();
ss.channels = /*provider->GetChannels()*/1;
pa_channel_map map;
pa_channel_map_init_auto(&map, ss.channels, PA_CHANNEL_MAP_DEFAULT);
@ -308,7 +308,7 @@ void PulseAudioPlayer::pa_stream_write(pa_stream *p, size_t length, PulseAudioPl
unsigned long maxframes = thread->end_frame - thread->cur_frame;
if (frames > maxframes) frames = maxframes;
void *buf = malloc(frames * bpf);
thread->provider->GetAudioWithVolume(buf, thread->cur_frame, frames, thread->volume);
thread->provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf), thread->cur_frame, frames, thread->volume);
::pa_stream_write(p, buf, frames*bpf, free, 0, PA_SEEK_RELATIVE);
thread->cur_frame += frames;
}
@ -324,4 +324,4 @@ void PulseAudioPlayer::pa_stream_notify(pa_stream *p, PulseAudioPlayer *thread)
std::unique_ptr<AudioPlayer> CreatePulseAudioPlayer(agi::AudioProvider *provider, wxWindow *) {
return agi::make_unique<PulseAudioPlayer>(provider);
}
#endif // WITH_LIBPULSE
#endif // WITH_LIBPULSE

View file

@ -0,0 +1,694 @@
// Copyright (c) 2019, Qirui Wang
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// Aegisub Project http://www.aegisub.org/
#ifdef WITH_XAUDIO2
#include "include/aegisub/audio_player.h"
#include "options.h"
#include <libaegisub/audio/provider.h>
#include <libaegisub/scoped_ptr.h>
#include <libaegisub/log.h>
#include <libaegisub/make_unique.h>
#ifndef XAUDIO2_REDIST
#include <xaudio2.h>
#else
#include <xaudio2redist.h>
#endif
namespace {
class XAudio2Thread;
/// @class XAudio2Player
/// @brief XAudio2-based audio player
///
/// The core design idea is to have a playback thread that performs all playback operations, and use the player object as a proxy to send commands to the playback thread.
class XAudio2Player final : public AudioPlayer {
/// The playback thread
std::unique_ptr<XAudio2Thread> thread;
/// Desired length in milliseconds to write ahead of the playback cursor
int WantedLatency;
/// Multiplier for WantedLatency to get total buffer length
int BufferLength;
/// @brief Tell whether playback thread is alive
/// @return True if there is a playback thread and it's ready
bool IsThreadAlive();
public:
/// @brief Constructor
XAudio2Player(agi::AudioProvider* provider);
/// @brief Destructor
~XAudio2Player() = default;
/// @brief Start playback
/// @param start First audio frame to play
/// @param count Number of audio frames to play
void Play(int64_t start, int64_t count);
/// @brief Stop audio playback
/// @param timerToo Whether to also stop the playback update timer
void Stop();
/// @brief Tell whether playback is active
/// @return True if audio is playing back
bool IsPlaying();
/// @brief Get playback end position
/// @return Audio frame index
///
/// Returns 0 if playback is stopped or there is no playback thread
int64_t GetEndPosition();
/// @brief Get approximate playback position
/// @return Index of audio frame user is currently hearing
///
/// Returns 0 if playback is stopped or there is no playback thread
int64_t GetCurrentPosition();
/// @brief Change playback end position
/// @param pos New end position
void SetEndPosition(int64_t pos);
/// @brief Change playback volume
/// @param vol Amplification factor
void SetVolume(double vol);
};
/// @brief RAII support class to init and de-init the COM library
struct COMInitialization {
/// Flag set if an inited COM library is managed
bool inited = false;
/// @brief Destructor, de-inits COM if it is inited
~COMInitialization() {
if (inited) CoUninitialize();
}
/// @brief Initialise the COM library as single-threaded apartment if isn't already inited by us
bool Init() {
if (!inited && SUCCEEDED(CoInitialize(nullptr)))
inited = true;
return inited;
}
};
struct ReleaseCOMObject {
void operator()(IUnknown* obj) {
if (obj) obj->Release();
}
};
/// @brief RAII wrapper around Win32 HANDLE type
struct Win32KernelHandle final : public agi::scoped_holder<HANDLE, BOOL(__stdcall*)(HANDLE)> {
/// @brief Create with a managed handle
/// @param handle Win32 handle to manage
Win32KernelHandle(HANDLE handle = 0) :scoped_holder(handle, CloseHandle) {}
Win32KernelHandle& operator=(HANDLE new_handle) {
scoped_holder::operator=(new_handle);
return *this;
}
};
/// @class XAudio2Thread
/// @brief Playback thread class for XAudio2Player
///
/// Not based on wxThread, but uses Win32 threads directly
class XAudio2Thread :public IXAudio2VoiceCallback {
/// @brief Win32 thread entry point
/// @param parameter Pointer to our thread object
/// @return Thread return value, always 0 here
static unsigned int __stdcall ThreadProc(void* parameter);
/// @brief Thread entry point
void Run();
/// @brief Check for error state and throw exception if one occurred
void CheckError();
/// Win32 handle to the thread
Win32KernelHandle thread_handle;
/// Event object, world to thread, set to start playback
Win32KernelHandle event_start_playback;
/// Event object, world to thread, set to stop playback
Win32KernelHandle event_stop_playback;
/// Event object, world to thread, set if playback end time was updated
Win32KernelHandle event_update_end_time;
/// Event object, world to thread, set if the volume was changed
Win32KernelHandle event_set_volume;
/// Event object, world to thread, set if the thread should end as soon as possible
Win32KernelHandle event_buffer_end;
/// Event object, world to thread, set if the thread should end as soon as possible
Win32KernelHandle event_kill_self;
/// Event object, thread to world, set when the thread has entered its main loop
Win32KernelHandle thread_running;
/// Event object, thread to world, set when playback is ongoing
Win32KernelHandle is_playing;
/// Event object, thread to world, set if an error state has occurred (implies thread is dying)
Win32KernelHandle error_happened;
/// Statically allocated error message text describing reason for error_happened being set
const char* error_message = nullptr;
/// Playback volume, 1.0 is "unchanged"
double volume = 1.0;
/// Audio frame to start playback at
int64_t start_frame = 0;
/// Audio frame to end playback at
int64_t end_frame = 0;
/// Desired length in milliseconds to write ahead of the playback cursor
int wanted_latency;
/// Multiplier for WantedLatency to get total buffer length
int buffer_length;
/// System millisecond timestamp of last playback start, used to calculate playback position
ULONGLONG last_playback_restart;
/// Audio provider to take sample data from
agi::AudioProvider* provider;
/// Buffer occupied indicator
std::vector<bool> buffer_occupied;
public:
/// @brief Constructor, creates and starts playback thread
/// @param provider Audio provider to take sample data from
/// @param WantedLatency Desired length in milliseconds to write ahead of the playback cursor
/// @param BufferLength Multiplier for WantedLatency to get total buffer length
XAudio2Thread(agi::AudioProvider* provider, int WantedLatency, int BufferLength);
/// @brief Destructor, waits for thread to have died
~XAudio2Thread();
// IXAudio2VoiceCallback
void STDMETHODCALLTYPE OnVoiceProcessingPassStart(UINT32 BytesRequired) override {}
void STDMETHODCALLTYPE OnVoiceProcessingPassEnd() override {}
void STDMETHODCALLTYPE OnStreamEnd() override {}
void STDMETHODCALLTYPE OnBufferStart(void* pBufferContext) override {}
void STDMETHODCALLTYPE OnBufferEnd(void* pBufferContext) override {
intptr_t i = reinterpret_cast<intptr_t>(pBufferContext);
buffer_occupied[i] = false;
SetEvent(event_buffer_end);
}
void STDMETHODCALLTYPE OnLoopEnd(void* pBufferContext) override {}
void STDMETHODCALLTYPE OnVoiceError(void* pBufferContext, HRESULT Error) override {}
/// @brief Start audio playback
/// @param start Audio frame to start playback at
/// @param count Number of audio frames to play
void Play(int64_t start, int64_t count);
/// @brief Stop audio playback
void Stop();
/// @brief Change audio playback end point
/// @param new_end_frame New last audio frame to play
///
/// Playback stops instantly if new_end_frame is before the current playback position
void SetEndFrame(int64_t new_end_frame);
/// @brief Change audio playback volume
/// @param new_volume New playback amplification factor, 1.0 is "unchanged"
void SetVolume(double new_volume);
/// @brief Tell whether audio playback is active
/// @return True if audio is being played back, false if it is not
bool IsPlaying();
/// @brief Get approximate current audio frame being heard by the user
/// @return Audio frame index
///
/// Returns 0 if not playing
int64_t GetCurrentFrame();
/// @brief Get audio playback end point
/// @return Audio frame index
int64_t GetEndFrame();
/// @brief Tell whether playback thread has died
/// @return True if thread is no longer running
bool IsDead();
};
unsigned int __stdcall XAudio2Thread::ThreadProc(void* parameter) {
static_cast<XAudio2Thread*>(parameter)->Run();
return 0;
}
/// Macro used to set error_message, error_happened and end the thread
#define REPORT_ERROR(msg) \
{ \
ResetEvent(is_playing); \
error_message = "XAudio2Thread: " msg; \
SetEvent(error_happened); \
return; \
}
void XAudio2Thread::Run() {
COMInitialization COM_library;
if (!COM_library.Init()) {
REPORT_ERROR("Could not initialise COM")
}
IXAudio2* pXAudio2;
IXAudio2SourceVoice* pSourceVoice;
HRESULT hr;
if (FAILED(hr = XAudio2Create(&pXAudio2, 0, XAUDIO2_DEFAULT_PROCESSOR))) {
REPORT_ERROR("Failed initializing XAudio2")
}
IXAudio2MasteringVoice* pMasterVoice = NULL;
if (FAILED(hr = pXAudio2->CreateMasteringVoice(&pMasterVoice))) {
REPORT_ERROR("Failed initializing XAudio2 MasteringVoice")
}
// Describe the wave format
WAVEFORMATEX wfx;
wfx.nSamplesPerSec = provider->GetSampleRate();
wfx.cbSize = 0;
bool original = true;
wfx.wFormatTag = provider->AreSamplesFloat() ? WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM;
wfx.nChannels = provider->GetChannels();
wfx.wBitsPerSample = provider->GetBytesPerSample() * 8;
wfx.nBlockAlign = wfx.nChannels * wfx.wBitsPerSample / 8;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
if (FAILED(hr = pXAudio2->CreateSourceVoice(&pSourceVoice, &wfx, 0, 2, this))) {
if (hr == XAUDIO2_E_INVALID_CALL) {
// Retry with 16bit mono
original = false;
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = 1;
wfx.wBitsPerSample = sizeof(int16_t) * 8;
wfx.nBlockAlign = wfx.nChannels * wfx.wBitsPerSample / 8;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
if (FAILED(hr = pXAudio2->CreateSourceVoice(&pSourceVoice, &wfx, 0, 2, this))) {
REPORT_ERROR("Failed initializing XAudio2 SourceVoice")
}
}
else {
REPORT_ERROR("Failed initializing XAudio2 SourceVoice")
}
}
// Now we're ready to roll!
SetEvent(thread_running);
bool running = true;
HANDLE events_to_wait[] = {
event_start_playback,
event_stop_playback,
event_update_end_time,
event_set_volume,
event_buffer_end,
event_kill_self
};
int64_t next_input_frame = 0;
DWORD buffer_offset = 0;
bool playback_should_be_running = false;
int current_latency = wanted_latency;
const int wanted_frames = wanted_latency * wfx.nSamplesPerSec / 1000;
const DWORD wanted_latency_bytes = wanted_frames * wfx.nBlockAlign;
std::vector<std::vector<BYTE> > buff(buffer_length);
for (auto& i : buff)
i.resize(wanted_latency_bytes);
while (running) {
DWORD wait_result = WaitForMultipleObjects(sizeof(events_to_wait) / sizeof(HANDLE), events_to_wait, FALSE, INFINITE);
switch (wait_result) {
case WAIT_OBJECT_0 + 0:
// Start or restart playback
pSourceVoice->Stop();
pSourceVoice->FlushSourceBuffers();
next_input_frame = start_frame;
playback_should_be_running = true;
pSourceVoice->Start();
SetEvent(is_playing);
goto do_fill_buffer;
case WAIT_OBJECT_0 + 1:
stop_playback:
// Stop playing
ResetEvent(is_playing);
pSourceVoice->Stop();
pSourceVoice->FlushSourceBuffers();
playback_should_be_running = false;
break;
case WAIT_OBJECT_0 + 2:
// Set end frame
if (end_frame <= next_input_frame)
goto stop_playback;
goto do_fill_buffer;
case WAIT_OBJECT_0 + 3:
// Change volume
pSourceVoice->SetVolume(volume);
break;
case WAIT_OBJECT_0 + 4:
// Buffer end
do_fill_buffer:
// Time to fill more into buffer
if (!playback_should_be_running)
break;
for (int i = 0; i < buffer_length; ++i) {
if (!buffer_occupied[i]) {
int fill_len = std::min<int>(end_frame - next_input_frame, wanted_frames);
if (fill_len <= 0)
break;
buffer_occupied[i] = true;
if (original)
provider->GetAudio(buff[i].data(), next_input_frame, fill_len);
else
provider->GetInt16MonoAudio(reinterpret_cast<int16_t*>(buff[i].data()), next_input_frame, fill_len);
next_input_frame += fill_len;
XAUDIO2_BUFFER xbf;
xbf.Flags = fill_len + next_input_frame == end_frame ? XAUDIO2_END_OF_STREAM : 0;
xbf.AudioBytes = fill_len * wfx.nBlockAlign;
xbf.pAudioData = buff[i].data();
xbf.PlayBegin = 0;
xbf.PlayLength = 0;
xbf.LoopBegin = 0;
xbf.LoopLength = 0;
xbf.LoopCount = 0;
xbf.pContext = reinterpret_cast<void*>(static_cast<intptr_t>(i));
if (FAILED(hr = pSourceVoice->SubmitSourceBuffer(&xbf))) {
REPORT_ERROR("Failed initializing Submit Buffer")
}
}
}