Merge branch 'xa2-ds' into feature
This commit is contained in:
arch1t3cht
commit
e7e87f5584
22 changed files with 981 additions and 206 deletions
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@ -21,13 +21,107 @@
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#include "libaegisub/log.h"
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#include "libaegisub/util.h"
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namespace agi {
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void AudioProvider::GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const {
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GetAudio(buf, start, count);
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namespace {
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template<typename Source>
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class ConvertFloatToInt16 {
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Source* src;
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public:
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ConvertFloatToInt16(Source* src) :src(src) {}
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int16_t operator[](size_t idx) const {
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Source expanded = src[idx] * 32768;
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return expanded < -32768 ? -32768 :
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expanded > 32767 ? 32767 :
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static_cast<int16_t>(expanded);
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}
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};
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// 8 bits per sample is assumed to be unsigned with a bias of 128,
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// while everything else is assumed to be signed with zero bias
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class ConvertIntToInt16 {
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void* src;
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int bytes_per_sample;
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public:
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ConvertIntToInt16(void* src, int bytes_per_sample) :src(src), bytes_per_sample(bytes_per_sample) {}
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const int16_t& operator[](size_t idx) const {
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return *reinterpret_cast<int16_t*>(reinterpret_cast<char*>(src) + (idx + 1) * bytes_per_sample - sizeof(int16_t));
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}
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};
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class ConvertUInt8ToInt16 {
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uint8_t* src;
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public:
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ConvertUInt8ToInt16(uint8_t* src) :src(src) {}
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int16_t operator[](size_t idx) const {
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return int16_t(src[idx]-128) << 8;
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}
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};
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template<typename Source>
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class DownmixToMono {
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Source src;
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int channels;
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public:
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DownmixToMono(Source src, int channels) :src(src), channels(channels) {}
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int16_t operator[](size_t idx) const {
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int ret = 0;
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// Just average the channels together
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for (int i = 0; i < channels; ++i)
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ret += src[idx * channels + i];
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return ret / channels;
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}
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};
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}
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namespace agi {
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void AudioProvider::FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const {
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if (!float_samples && bytes_per_sample == 2 && channels == 1) {
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FillBuffer(buf, start, count);
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return;
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}
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void* buff = malloc(bytes_per_sample * count * channels);
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FillBuffer(buff, start, count);
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if (channels == 1) {
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if (float_samples) {
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if (bytes_per_sample == sizeof(float))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff))[i];
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else if (bytes_per_sample == sizeof(double))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff))[i];
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}
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else {
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if (bytes_per_sample == sizeof(uint8_t))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff))[i];
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else
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for (int64_t i = 0; i < count; ++i)
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buf[i] = ConvertIntToInt16(buff, bytes_per_sample)[i];
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}
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}
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else {
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if (float_samples) {
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if (bytes_per_sample == sizeof(float))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = DownmixToMono<ConvertFloatToInt16<float> >(ConvertFloatToInt16<float>(reinterpret_cast<float*>(buff)), channels)[i];
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else if (bytes_per_sample == sizeof(double))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = DownmixToMono<ConvertFloatToInt16<double> >(ConvertFloatToInt16<double>(reinterpret_cast<double*>(buff)), channels)[i];
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}
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else {
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if (bytes_per_sample == sizeof(uint8_t))
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for (int64_t i = 0; i < count; ++i)
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buf[i] = DownmixToMono<ConvertUInt8ToInt16>(ConvertUInt8ToInt16(reinterpret_cast<uint8_t*>(buff)), channels)[i];
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else
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for (int64_t i = 0; i < count; ++i)
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buf[i] = DownmixToMono<ConvertIntToInt16>(ConvertIntToInt16(buff, bytes_per_sample), channels)[i];
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}
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}
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free(buff);
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}
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void AudioProvider::GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const {
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GetInt16MonoAudio(buf, start, count);
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if (volume == 1.0) return;
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if (bytes_per_sample != 2)
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throw agi::InternalError("GetAudioWithVolume called on unconverted audio stream");
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auto buffer = static_cast<int16_t *>(buf);
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for (size_t i = 0; i < (size_t)count; ++i)
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@ -75,6 +169,39 @@ void AudioProvider::GetAudio(void *buf, int64_t start, int64_t count) const {
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}
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}
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void AudioProvider::GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const {
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if (start < 0) {
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memset(buf, 0, sizeof(int16_t) * std::min(-start, count));
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buf -= start;
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count += start;
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start = 0;
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}
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if (start + count > num_samples) {
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int64_t zero_count = std::min(count, start + count - num_samples);
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count -= zero_count;
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memset(buf + count, 0, sizeof(int16_t) * zero_count);
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}
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if (count <= 0) return;
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try {
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FillBufferInt16Mono(buf, start, count);
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}
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catch (AudioDecodeError const& e) {
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// We don't have any good way to report errors here, so just log the
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// failure and return silence
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LOG_E("audio_provider") << e.GetMessage();
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memset(buf, 0, sizeof(int16_t) * count);
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return;
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}
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catch (...) {
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LOG_E("audio_provider") << "Unknown audio decoding error";
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memset(buf, 0, sizeof(int16_t) * count);
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return;
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}
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}
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namespace {
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class writer {
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io::Save outfile;
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@ -114,7 +241,7 @@ void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int star
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out.write("WAVEfmt ");
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out.write<int32_t>(16); // Size of chunk
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out.write<int16_t>(1); // compression format (PCM)
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out.write<int16_t>(provider.AreSamplesFloat() ? 3 : 1); // compression format (1: WAVE_FORMAT_PCM, 3: WAVE_FORMAT_IEEE_FLOAT)
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out.write<int16_t>(provider.GetChannels());
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out.write<int32_t>(provider.GetSampleRate());
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out.write<int32_t>(provider.GetSampleRate() * provider.GetChannels() * provider.GetBytesPerSample());
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@ -22,119 +22,19 @@
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#include <limits>
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using namespace agi;
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/// Anything integral -> 16 bit signed machine-endian audio converter
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namespace {
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template<class Target>
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class BitdepthConvertAudioProvider final : public AudioProviderWrapper {
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int src_bytes_per_sample;
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mutable std::vector<uint8_t> src_buf;
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class ConvertAudioProvider final : public AudioProviderWrapper {
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public:
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BitdepthConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
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if (bytes_per_sample > 8)
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throw AudioProviderError("Audio format converter: audio with bitdepths greater than 64 bits/sample is currently unsupported");
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src_bytes_per_sample = bytes_per_sample;
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bytes_per_sample = sizeof(Target);
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}
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * src_bytes_per_sample * channels);
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source->GetAudio(src_buf.data(), start, count);
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auto dest = static_cast<int16_t*>(buf);
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for (int64_t i = 0; i < count * channels; ++i) {
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int64_t sample = 0;
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// 8 bits per sample is assumed to be unsigned with a bias of 127,
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// while everything else is assumed to be signed with zero bias
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if (src_bytes_per_sample == 1)
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sample = src_buf[i] - 128;
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else {
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for (int j = src_bytes_per_sample; j > 0; --j) {
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sample <<= 8;
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sample += src_buf[i * src_bytes_per_sample + j - 1];
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}
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}
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if (static_cast<size_t>(src_bytes_per_sample) > sizeof(Target))
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sample /= 1LL << (src_bytes_per_sample - sizeof(Target)) * 8;
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else if (static_cast<size_t>(src_bytes_per_sample) < sizeof(Target))
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sample *= 1LL << (sizeof(Target) - src_bytes_per_sample ) * 8;
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dest[i] = static_cast<Target>(sample);
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}
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}
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};
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/// Floating point -> 16 bit signed machine-endian audio converter
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template<class Source, class Target>
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class FloatConvertAudioProvider final : public AudioProviderWrapper {
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mutable std::vector<Source> src_buf;
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public:
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FloatConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
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bytes_per_sample = sizeof(Target);
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ConvertAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
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float_samples = false;
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}
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * channels);
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source->GetAudio(&src_buf[0], start, count);
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auto dest = static_cast<Target*>(buf);
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for (size_t i = 0; i < static_cast<size_t>(count * channels); ++i) {
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Source expanded;
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if (src_buf[i] < 0)
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expanded = static_cast<Target>(-src_buf[i] * std::numeric_limits<Target>::min());
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else
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expanded = static_cast<Target>(src_buf[i] * std::numeric_limits<Target>::max());
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dest[i] = expanded < std::numeric_limits<Target>::min() ? std::numeric_limits<Target>::min() :
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expanded > std::numeric_limits<Target>::max() ? std::numeric_limits<Target>::max() :
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static_cast<Target>(expanded);
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}
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}
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};
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/// Non-mono 16-bit signed machine-endian -> mono 16-bit signed machine endian converter
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class DownmixAudioProvider final : public AudioProviderWrapper {
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int src_channels;
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mutable std::vector<int16_t> src_buf;
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public:
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DownmixAudioProvider(std::unique_ptr<AudioProvider> src) : AudioProviderWrapper(std::move(src)) {
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src_channels = channels;
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channels = 1;
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bytes_per_sample = sizeof(int16_t);
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}
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void FillBuffer(void *buf, int64_t start, int64_t count64) const override {
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auto count = static_cast<size_t>(count64);
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assert(count == count64);
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src_buf.resize(count * src_channels);
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source->GetAudio(&src_buf[0], start, count);
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auto dst = static_cast<int16_t*>(buf);
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// Just average the channels together
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while (count-- > 0) {
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int sum = 0;
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for (int c = 0; c < src_channels; ++c)
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sum += src_buf[count * src_channels + c];
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dst[count] = static_cast<int16_t>(sum / src_channels);
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}
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void FillBuffer(void *buf, int64_t start, int64_t count) const override {
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source->GetInt16MonoAudio(reinterpret_cast<int16_t*>(buf), start, count);
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}
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};
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/// Sample doubler with linear interpolation for the samples provider
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/// Requires 16-bit mono input
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class SampleDoublingAudioProvider final : public AudioProviderWrapper {
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@ -177,27 +77,24 @@ std::unique_ptr<AudioProvider> CreateConvertAudioProvider(std::unique_ptr<AudioP
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// Ensure 16-bit audio with proper endianness
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if (provider->AreSamplesFloat()) {
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LOG_D("audio_provider") << "Converting float to S16";
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if (provider->GetBytesPerSample() == sizeof(float))
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provider = agi::make_unique<FloatConvertAudioProvider<float, int16_t>>(std::move(provider));
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else
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provider = agi::make_unique<FloatConvertAudioProvider<double, int16_t>>(std::move(provider));
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}
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if (provider->GetBytesPerSample() != 2) {
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LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample or wrong endian to S16";
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provider = agi::make_unique<BitdepthConvertAudioProvider<int16_t>>(std::move(provider));
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LOG_D("audio_provider") << "Converting " << provider->GetBytesPerSample() << " bytes per sample to S16";
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}
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// We currently only support mono audio
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if (provider->GetChannels() != 1) {
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LOG_D("audio_provider") << "Downmixing to mono from " << provider->GetChannels() << " channels";
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provider = agi::make_unique<DownmixAudioProvider>(std::move(provider));
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}
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// Some players don't like low sample rate audio
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if (provider->GetSampleRate() < 32000) {
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provider = agi::make_unique<ConvertAudioProvider>(std::move(provider));
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while (provider->GetSampleRate() < 32000) {
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LOG_D("audio_provider") << "Doubling sample rate";
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provider = agi::make_unique<SampleDoublingAudioProvider>(std::move(provider));
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}
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}
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return provider;
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}
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@ -43,15 +43,15 @@ class HDAudioProvider final : public AudioProviderWrapper {
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}
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if (count > 0) {
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start *= bytes_per_sample;
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count *= bytes_per_sample;
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start *= bytes_per_sample * channels;
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count *= bytes_per_sample * channels;
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memcpy(buf, file.read(start, count), count);
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}
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}
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fs::path CacheFilename(fs::path const& dir) {
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// Check free space
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if ((uint64_t)num_samples * bytes_per_sample > fs::FreeSpace(dir))
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if ((uint64_t)num_samples * bytes_per_sample * channels > fs::FreeSpace(dir))
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throw AudioProviderError("Not enough free disk space in " + dir.string() + " to cache the audio");
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return format("audio-%lld-%lld", time(nullptr),
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@ -61,7 +61,7 @@ class HDAudioProvider final : public AudioProviderWrapper {
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public:
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HDAudioProvider(std::unique_ptr<AudioProvider> src, agi::fs::path const& dir)
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: AudioProviderWrapper(std::move(src))
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, file(dir / CacheFilename(dir), num_samples * bytes_per_sample)
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, file(dir / CacheFilename(dir), num_samples * bytes_per_sample * channels)
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{
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decoded_samples = 0;
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decoder = std::thread([&] {
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|
|
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@ -29,6 +29,11 @@ class LockAudioProvider final : public agi::AudioProviderWrapper {
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source->GetAudio(buf, start, count);
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}
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void FillBufferInt16Mono(int16_t *buf, int64_t start, int64_t count) const override {
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std::unique_lock<std::mutex> lock(mutex);
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source->GetInt16MonoAudio(buf, start, count);
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}
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public:
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LockAudioProvider(std::unique_ptr<AudioProvider> src)
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: AudioProviderWrapper(std::move(src))
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@ -46,14 +46,14 @@ public:
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decoded_samples = 0;
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try {
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blockcache.resize((source->GetNumSamples() * source->GetBytesPerSample() + CacheBlockSize - 1) >> CacheBits);
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blockcache.resize((num_samples * bytes_per_sample * channels + CacheBlockSize - 1) >> CacheBits);
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}
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catch (std::bad_alloc const&) {
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throw AudioProviderError("Not enough memory available to cache in RAM");
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}
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decoder = std::thread([&] {
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int64_t readsize = CacheBlockSize / source->GetBytesPerSample();
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int64_t readsize = CacheBlockSize / bytes_per_sample / channels;
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for (size_t i = 0; i < blockcache.size(); i++) {
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if (cancelled) break;
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auto actual_read = std::min<int64_t>(readsize, num_samples - i * readsize);
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@ -71,20 +71,22 @@ public:
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void RAMAudioProvider::FillBuffer(void *buf, int64_t start, int64_t count) const {
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auto charbuf = static_cast<char *>(buf);
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for (int64_t bytes_remaining = count * bytes_per_sample; bytes_remaining; ) {
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for (int64_t bytes_remaining = count * bytes_per_sample * channels; bytes_remaining; ) {
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if (start >= decoded_samples) {
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memset(charbuf, 0, bytes_remaining);
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break;
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}
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const int i = (start * bytes_per_sample) >> CacheBits;
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const int start_offset = (start * bytes_per_sample) & (CacheBlockSize-1);
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const int read_size = std::min<int>(bytes_remaining, CacheBlockSize - start_offset);
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const int64_t samples_per_block = CacheBlockSize / bytes_per_sample / channels;
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const size_t i = start / samples_per_block;
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const int start_offset = (start % samples_per_block) * bytes_per_sample * channels;
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const int read_size = std::min<int>(bytes_remaining, samples_per_block * bytes_per_sample * channels - start_offset);
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|
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memcpy(charbuf, &blockcache[i][start_offset], read_size);
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charbuf += read_size;
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bytes_remaining -= read_size;
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start += read_size / bytes_per_sample;
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start += read_size / bytes_per_sample / channels;
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}
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}
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}
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@ -20,8 +20,8 @@
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|||
#include <libaegisub/fs_fwd.h>
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||||
|
||||
#include <atomic>
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||||
#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; }
|
||||
|
|
|
|||
28
meson.build
28
meson.build
|
|
@ -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')
|
||||
|
|
@ -266,7 +266,8 @@ if get_option('vapoursynth').enabled()
|
|||
dep_avail += 'VapourSynth'
|
||||
endif
|
||||
|
||||
if host_machine.system() == 'windows' and not get_option('directsound').disabled()
|
||||
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'))
|
||||
|
|
@ -283,6 +284,29 @@ if host_machine.system() == 'windows' and not get_option('directsound').disabled
|
|||
endif
|
||||
endif
|
||||
|
||||
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
|
||||
|
||||
if host_machine.system() == 'darwin'
|
||||
frameworks_dep = dependency('appleframeworks', modules : ['CoreText', 'CoreFoundation', 'AppKit', 'Carbon', 'IOKit'])
|
||||
deps += frameworks_dep
|
||||
|
|
|
|||
|
|
@ -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')
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
|
|||
|
|
@ -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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -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();
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -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);
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
|
|
|
|||
|
|
@ -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;
|
||||
}
|
||||
|
|
|
|||
694
src/audio_player_xaudio2.cpp
Normal file
694
src/audio_player_xaudio2.cpp
Normal 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")
|
||||
}
|
||||
}
|
||||
}
|
||||
break;
|
||||
|
||||
case WAIT_OBJECT_0 + 5:
|
||||
// Perform suicide
|
||||
running = false;
|
||||
pXAudio2->Release();
|
||||
ResetEvent(is_playing);
|
||||
playback_should_be_running = false;
|
||||
break;
|
||||
|
||||
default:
|
||||
REPORT_ERROR("Something bad happened while waiting on events in playback loop, either the wait failed or an event object was abandoned.")
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
#undef REPORT_ERROR
|
||||
|
||||
void XAudio2Thread::CheckError()
|
||||
{
|
||||
try {
|
||||
switch (WaitForSingleObject(error_happened, 0))
|
||||
{
|
||||
case WAIT_OBJECT_0:
|
||||
throw error_message;
|
||||
|
||||
case WAIT_ABANDONED:
|
||||
throw "The XAudio2Thread error signal event was abandoned, somehow. This should not happen.";
|
||||
|
||||
case WAIT_FAILED:
|
||||
throw "Failed checking state of XAudio2Thread error signal event.";
|
||||
|
||||
case WAIT_TIMEOUT:
|
||||
default:
|
||||
return;
|
||||
}
|
||||
}
|
||||
catch (...) {
|
||||
ResetEvent(is_playing);
|
||||
ResetEvent(thread_running);
|
||||
throw;
|
||||
}
|
||||
}
|
||||
|
||||
XAudio2Thread::XAudio2Thread(agi::AudioProvider* provider, int WantedLatency, int BufferLength)
|
||||
: event_start_playback(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, event_stop_playback(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, event_update_end_time(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, event_set_volume(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, event_buffer_end(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, event_kill_self(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, thread_running(CreateEvent(0, TRUE, FALSE, 0))
|
||||
, is_playing(CreateEvent(0, TRUE, FALSE, 0))
|
||||
, error_happened(CreateEvent(0, FALSE, FALSE, 0))
|
||||
, wanted_latency(WantedLatency)
|
||||
, buffer_length(BufferLength < XAUDIO2_MAX_QUEUED_BUFFERS ? BufferLength : XAUDIO2_MAX_QUEUED_BUFFERS)
|
||||
, provider(provider)
|
||||
, buffer_occupied(BufferLength)
|
||||
{
|
||||
if (!(thread_handle = (HANDLE)_beginthreadex(0, 0, ThreadProc, this, 0, 0))) {
|
||||
throw AudioPlayerOpenError("Failed creating playback thread in XAudio2Player. This is bad.");
|
||||
}
|
||||
|
||||
HANDLE running_or_error[] = { thread_running, error_happened };
|
||||
switch (WaitForMultipleObjects(2, running_or_error, FALSE, INFINITE)) {
|
||||
case WAIT_OBJECT_0:
|
||||
// running, all good
|
||||
return;
|
||||
|
||||
case WAIT_OBJECT_0 + 1:
|
||||
// error happened, we fail
|
||||
throw AudioPlayerOpenError(error_message ? error_message : "Failed wait for thread start or thread error in XAudio2Player. This is bad.");
|
||||
|
||||
default:
|
||||
throw AudioPlayerOpenError("Failed wait for thread start or thread error in XAudio2Player. This is bad.");
|
||||
}
|
||||
}
|
||||
|
||||
XAudio2Thread::~XAudio2Thread() {
|
||||
SetEvent(event_kill_self);
|
||||
WaitForSingleObject(thread_handle, INFINITE);
|
||||
}
|
||||
|
||||
void XAudio2Thread::Play(int64_t start, int64_t count)
|
||||
{
|
||||
CheckError();
|
||||
|
||||
start_frame = start;
|
||||
end_frame = start + count;
|
||||
SetEvent(event_start_playback);
|
||||
|
||||
last_playback_restart = GetTickCount64();
|
||||
|
||||
// Block until playback actually begins to avoid race conditions with
|
||||
// checking if playback is in progress
|
||||
HANDLE events_to_wait[] = { is_playing, error_happened };
|
||||
switch (WaitForMultipleObjects(2, events_to_wait, FALSE, INFINITE)) {
|
||||
case WAIT_OBJECT_0 + 0: // Playing
|
||||
LOG_D("audio/player/xaudio2") << "Playback begun";
|
||||
break;
|
||||
case WAIT_OBJECT_0 + 1: // Error
|
||||
throw error_message;
|
||||
default:
|
||||
throw agi::InternalError("Unexpected result from WaitForMultipleObjects in XAudio2Thread::Play");
|
||||
}
|
||||
}
|
||||
|
||||
void XAudio2Thread::Stop() {
|
||||
CheckError();
|
||||
|
||||
SetEvent(event_stop_playback);
|
||||
}
|
||||
|
||||
void XAudio2Thread::SetEndFrame(int64_t new_end_frame) {
|
||||
CheckError();
|
||||
|
||||
end_frame = new_end_frame;
|
||||
SetEvent(event_update_end_time);
|
||||
}
|
||||
|
||||
void XAudio2Thread::SetVolume(double new_volume) {
|
||||
CheckError();
|
||||
|
||||
volume = new_volume;
|
||||
SetEvent(event_set_volume);
|
||||
}
|
||||
|
||||
bool XAudio2Thread::IsPlaying() {
|
||||
CheckError();
|
||||
|
||||
switch (WaitForSingleObject(is_playing, 0))
|
||||
{
|
||||
case WAIT_ABANDONED:
|
||||
throw "The XAudio2Thread playback state event was abandoned, somehow. This should not happen.";
|
||||
|
||||
case WAIT_FAILED:
|
||||
throw "Failed checking state of XAudio2Thread playback state event.";
|
||||
|
||||
case WAIT_OBJECT_0:
|
||||
return true;
|
||||
|
||||
case WAIT_TIMEOUT:
|
||||
default:
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
int64_t XAudio2Thread::GetCurrentFrame() {
|
||||
CheckError();
|
||||
if (!IsPlaying()) return 0;
|
||||
ULONGLONG milliseconds_elapsed = GetTickCount64() - last_playback_restart;
|
||||
return start_frame + milliseconds_elapsed * provider->GetSampleRate() / 1000;
|
||||
}
|
||||
|
||||
int64_t XAudio2Thread::GetEndFrame() {
|
||||
CheckError();
|
||||
return end_frame;
|
||||
}
|
||||
|
||||
bool XAudio2Thread::IsDead() {
|
||||
switch (WaitForSingleObject(thread_running, 0))
|
||||
{
|
||||
case WAIT_OBJECT_0:
|
||||
return false;
|
||||
default:
|
||||
return true;
|
||||
}
|
||||
}
|
||||
|
||||
XAudio2Player::XAudio2Player(agi::AudioProvider* provider) :AudioPlayer(provider) {
|
||||
// The buffer will hold BufferLength times WantedLatency milliseconds of audio
|
||||
WantedLatency = OPT_GET("Player/Audio/DirectSound/Buffer Latency")->GetInt();
|
||||
BufferLength = OPT_GET("Player/Audio/DirectSound/Buffer Length")->GetInt();
|
||||
|
||||
// sanity checking
|
||||
if (WantedLatency <= 0)
|
||||
WantedLatency = 100;
|
||||
if (BufferLength <= 0)
|
||||
BufferLength = 5;
|
||||
|
||||
try {
|
||||
thread = agi::make_unique<XAudio2Thread>(provider, WantedLatency, BufferLength);
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
throw AudioPlayerOpenError(msg);
|
||||
}
|
||||
}
|
||||
|
||||
bool XAudio2Player::IsThreadAlive() {
|
||||
if (thread && thread->IsDead())
|
||||
thread.reset();
|
||||
return static_cast<bool>(thread);
|
||||
}
|
||||
|
||||
void XAudio2Player::Play(int64_t start, int64_t count) {
|
||||
try {
|
||||
thread->Play(start, count);
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
}
|
||||
}
|
||||
|
||||
void XAudio2Player::Stop() {
|
||||
try {
|
||||
if (IsThreadAlive()) thread->Stop();
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
}
|
||||
}
|
||||
|
||||
bool XAudio2Player::IsPlaying() {
|
||||
try {
|
||||
if (!IsThreadAlive()) return false;
|
||||
return thread->IsPlaying();
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
return false;
|
||||
}
|
||||
}
|
||||
|
||||
int64_t XAudio2Player::GetEndPosition() {
|
||||
try {
|
||||
if (!IsThreadAlive()) return 0;
|
||||
return thread->GetEndFrame();
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
int64_t XAudio2Player::GetCurrentPosition() {
|
||||
try {
|
||||
if (!IsThreadAlive()) return 0;
|
||||
return thread->GetCurrentFrame();
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
return 0;
|
||||
}
|
||||
}
|
||||
|
||||
void XAudio2Player::SetEndPosition(int64_t pos) {
|
||||
try {
|
||||
if (IsThreadAlive()) thread->SetEndFrame(pos);
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
}
|
||||
}
|
||||
|
||||
void XAudio2Player::SetVolume(double vol) {
|
||||
try {
|
||||
if (IsThreadAlive()) thread->SetVolume(vol);
|
||||
}
|
||||
catch (const char* msg) {
|
||||
LOG_E("audio/player/xaudio2") << msg;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
std::unique_ptr<AudioPlayer> CreateXAudio2Player(agi::AudioProvider* provider, wxWindow*) {
|
||||
return agi::make_unique<XAudio2Player>(provider);
|
||||
}
|
||||
|
||||
#endif // WITH_XAUDIO2
|
||||
|
|
@ -208,7 +208,7 @@ void AudioSpectrumRenderer::FillBlock(size_t block_index, float *block)
|
|||
assert(block);
|
||||
|
||||
int64_t first_sample = (((int64_t)block_index) << derivation_dist) - ((int64_t)1 << derivation_size);
|
||||
provider->GetAudio(&audio_scratch[0], first_sample, 2 << derivation_size);
|
||||
provider->GetInt16MonoAudio(audio_scratch.data(), first_sample, 2 << derivation_size);
|
||||
|
||||
// Because the FFTs used here are unnormalized DFTs, we have to compensate
|
||||
// the possible length difference between derivation_size used in the
|
||||
|
|
|
|||
|
|
@ -88,7 +88,7 @@ void AudioWaveformRenderer::Render(wxBitmap &bmp, int start, AudioRenderingStyle
|
|||
|
||||
for (int x = 0; x < rect.width; ++x)
|
||||
{
|
||||
provider->GetAudio(audio_buffer.get(), (int64_t)cur_sample, (int64_t)pixel_samples);
|
||||
provider->GetInt16MonoAudio(reinterpret_cast<int16_t*>(audio_buffer.get()), (int64_t)cur_sample, (int64_t)pixel_samples);
|
||||
cur_sample += pixel_samples;
|
||||
|
||||
int peak_min = 0, peak_max = 0;
|
||||
|
|
|
|||
|
|
@ -233,7 +233,7 @@ opt_src = [
|
|||
['OSS', 'audio_player_oss.cpp'],
|
||||
['DirectSound', ['audio_player_dsound.cpp',
|
||||
'audio_player_dsound2.cpp']],
|
||||
|
||||
['XAudio2', 'audio_player_xaudio2.cpp'],
|
||||
['FFMS2', ['audio_provider_ffmpegsource.cpp',
|
||||
'video_provider_ffmpegsource.cpp',
|
||||
'ffmpegsource_common.cpp']],
|
||||
|
|
|
|||
|
|
@ -39,8 +39,13 @@ eyedropper_cursor CURSOR "../bitmaps/windows/eyedropper.cur"
|
|||
#endif
|
||||
|
||||
VS_VERSION_INFO VERSIONINFO
|
||||
#ifdef TAGGED_RELEASE
|
||||
FILEVERSION RESOURCE_BASE_VERSION, BUILD_GIT_VERSION_NUMBER
|
||||
PRODUCTVERSION RESOURCE_BASE_VERSION, 0
|
||||
#else
|
||||
FILEVERSION BUILD_GIT_VERSION_NUMBER, BUILD_GIT_VERSION_NUMBER
|
||||
PRODUCTVERSION BUILD_GIT_VERSION_NUMBER, 0
|
||||
#endif
|
||||
FILEFLAGSMASK VS_FFI_FILEFLAGSMASK
|
||||
FILEFLAGS (AGI_RC_FLAG_DEBUG|AGI_RC_FLAG_PRERELEASE)
|
||||
FILEOS VOS__WINDOWS32
|
||||
|
|
|
|||
|
|
@ -172,21 +172,21 @@ TEST(lagi_audio, save_audio_clip_out_of_audio_range) {
|
|||
|
||||
TEST(lagi_audio, get_with_volume) {
|
||||
TestAudioProvider<> provider;
|
||||
uint16_t buff[4];
|
||||
int16_t buff[4];
|
||||
|
||||
provider.GetAudioWithVolume(buff, 0, 4, 1.0);
|
||||
provider.GetInt16MonoAudioWithVolume(buff, 0, 4, 1.0);
|
||||
EXPECT_EQ(0, buff[0]);
|
||||
EXPECT_EQ(1, buff[1]);
|
||||
EXPECT_EQ(2, buff[2]);
|
||||
EXPECT_EQ(3, buff[3]);
|
||||
|
||||
provider.GetAudioWithVolume(buff, 0, 4, 0.0);
|
||||
provider.GetInt16MonoAudioWithVolume(buff, 0, 4, 0.0);
|
||||
EXPECT_EQ(0, buff[0]);
|
||||
EXPECT_EQ(0, buff[1]);
|
||||
EXPECT_EQ(0, buff[2]);
|
||||
EXPECT_EQ(0, buff[3]);
|
||||
|
||||
provider.GetAudioWithVolume(buff, 0, 4, 2.0);
|
||||
provider.GetInt16MonoAudioWithVolume(buff, 0, 4, 2.0);
|
||||
EXPECT_EQ(0, buff[0]);
|
||||
EXPECT_EQ(2, buff[1]);
|
||||
EXPECT_EQ(4, buff[2]);
|
||||
|
|
@ -195,8 +195,8 @@ TEST(lagi_audio, get_with_volume) {
|
|||
|
||||
TEST(lagi_audio, volume_should_clamp_rather_than_wrap) {
|
||||
TestAudioProvider<> provider;
|
||||
uint16_t buff[1];
|
||||
provider.GetAudioWithVolume(buff, 30000, 1, 2.0);
|
||||
int16_t buff[1];
|
||||
provider.GetInt16MonoAudioWithVolume(buff, 30000, 1, 2.0);
|
||||
EXPECT_EQ(SHRT_MAX, buff[0]);
|
||||
}
|
||||
|
||||
|
|
@ -232,7 +232,7 @@ TEST(lagi_audio, convert_8bit) {
|
|||
auto provider = agi::CreateConvertAudioProvider(agi::make_unique<TestAudioProvider<uint8_t>>());
|
||||
|
||||
int16_t data[256];
|
||||
provider->GetAudio(data, 0, 256);
|
||||
provider->GetInt16MonoAudio(data, 0, 256);
|
||||
for (int i = 0; i < 256; ++i)
|
||||
ASSERT_EQ((i - 128) * 256, data[i]);
|
||||
}
|
||||
|
|
@ -243,13 +243,13 @@ TEST(lagi_audio, convert_32bit) {
|
|||
auto provider = agi::CreateConvertAudioProvider(std::move(src));
|
||||
|
||||
int16_t sample;
|
||||
provider->GetAudio(&sample, 0, 1);
|
||||
provider->GetInt16MonoAudio(&sample, 0, 1);
|
||||
EXPECT_EQ(SHRT_MIN, sample);
|
||||
|
||||
provider->GetAudio(&sample, 1LL << 31, 1);
|
||||
provider->GetInt16MonoAudio(&sample, 1LL << 31, 1);
|
||||
EXPECT_EQ(0, sample);
|
||||
|
||||
provider->GetAudio(&sample, (1LL << 32) - 1, 1);
|
||||
provider->GetInt16MonoAudio(&sample, (1LL << 32) - 1, 1);
|
||||
EXPECT_EQ(SHRT_MAX, sample);
|
||||
}
|
||||
|
||||
|
|
@ -310,10 +310,10 @@ TEST(lagi_audio, stereo_downmix) {
|
|||
};
|
||||
|
||||
auto provider = agi::CreateConvertAudioProvider(agi::make_unique<AudioProvider>());
|
||||
EXPECT_EQ(1, provider->GetChannels());
|
||||
EXPECT_EQ(2, provider->GetChannels());
|
||||
|
||||
int16_t samples[100];
|
||||
provider->GetAudio(samples, 0, 100);
|
||||
provider->GetInt16MonoAudio(samples, 0, 100);
|
||||
for (int i = 0; i < 100; ++i)
|
||||
EXPECT_EQ(i, samples[i]);
|
||||
}
|
||||
|
|
@ -333,27 +333,27 @@ struct FloatAudioProvider : agi::AudioProvider {
|
|||
auto out = static_cast<Float *>(buf);
|
||||
for (int64_t end = start + count; start < end; ++start) {
|
||||
auto shifted = start + SHRT_MIN;
|
||||
*out++ = (Float)(1.0 * shifted / (shifted < 0 ? -SHRT_MIN : SHRT_MAX));
|
||||
*out++ = (Float)(shifted) / (-SHRT_MIN);
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
TEST(lagi_audio, float_conversion) {
|
||||
auto provider = agi::CreateConvertAudioProvider(agi::make_unique<FloatAudioProvider<float>>());
|
||||
EXPECT_FALSE(provider->AreSamplesFloat());
|
||||
EXPECT_TRUE(provider->AreSamplesFloat());
|
||||
|
||||
int16_t samples[1 << 16];
|
||||
provider->GetAudio(samples, 0, 1 << 16);
|
||||
provider->GetInt16MonoAudio(samples, 0, 1 << 16);
|
||||
for (int i = 0; i < (1 << 16); ++i)
|
||||
ASSERT_EQ(i + SHRT_MIN, samples[i]);
|
||||
}
|
||||
|
||||
TEST(lagi_audio, double_conversion) {
|
||||
auto provider = agi::CreateConvertAudioProvider(agi::make_unique<FloatAudioProvider<double>>());
|
||||
EXPECT_FALSE(provider->AreSamplesFloat());
|
||||
EXPECT_TRUE(provider->AreSamplesFloat());
|
||||
|
||||
int16_t samples[1 << 16];
|
||||
provider->GetAudio(samples, 0, 1 << 16);
|
||||
provider->GetInt16MonoAudio(samples, 0, 1 << 16);
|
||||
for (int i = 0; i < (1 << 16); ++i)
|
||||
ASSERT_EQ(i + SHRT_MIN, samples[i]);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue