mirror of https://github.com/odrling/Aegisub
Implement wangqr Audio Changes:
- To allow for XAudio2 to work properly, we need to rework how does provider work since they only are used to be able to take in mono audio. - Other providers have been dumbed down to accept single channel audio since originally aegisub only accepted 1 channel audio. - meson.build has been modified to accommodate for xaudio, as we currently don't accept redistributable forms of xaudio, we need to work around the WinNT version. - There has been 1 more fix to res.rc to allow for compiling on non tagged releases.
This commit is contained in:
Ristellise
parent
3dfea0c315
commit
fd28458ed8
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@ -497,6 +497,7 @@ set_property(SOURCE src/subtitles_provider_csri.cpp PROPERTY INCLUDE_DIRECTORIES
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if(MSVC)
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target_link_libraries (Aegisub dsound)
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add_definitions("-DWITH_DIRECTSOUND")
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add_definitions("-DWITH_XAUDIO2")
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target_sources(Aegisub PRIVATE src/audio_player_dsound.cpp src/audio_player_dsound2.cpp src/audio_player_xaudio2.cpp)
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endif(MSVC)
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@ -21,13 +21,108 @@
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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 +170,39 @@ 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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@ -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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@ -177,26 +77,23 @@ 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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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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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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@ -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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@ -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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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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@ -20,7 +20,6 @@
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#include <libaegisub/fs_fwd.h>
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#include <atomic>
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#include <memory>
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#include <vector>
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namespace agi {
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@ -37,6 +36,7 @@ protected:
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bool float_samples = false;
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virtual void FillBuffer(void *buf, int64_t start, int64_t count) const = 0;
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virtual void FillBufferInt16Mono(int16_t* buf, int64_t start, int64_t count) const;
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void ZeroFill(void *buf, int64_t count) const;
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@ -44,7 +44,8 @@ public:
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virtual ~AudioProvider() = default;
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void GetAudio(void *buf, int64_t start, int64_t count) const;
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void GetAudioWithVolume(void *buf, int64_t start, int64_t count, double volume) const;
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void GetInt16MonoAudio(int16_t* buf, int64_t start, int64_t count) const;
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void GetInt16MonoAudioWithVolume(int16_t *buf, int64_t start, int64_t count, double volume) const;
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int64_t GetNumSamples() const { return num_samples; }
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int64_t GetDecodedSamples() const { return decoded_samples; }
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@ -93,4 +94,4 @@ std::unique_ptr<AudioProvider> CreateHDAudioProvider(std::unique_ptr<AudioProvid
|
|||
std::unique_ptr<AudioProvider> CreateRAMAudioProvider(std::unique_ptr<AudioProvider> source_provider);
|
||||
|
||||
void SaveAudioClip(AudioProvider const& provider, fs::path const& path, int start_time, int end_time);
|
||||
}
|
||||
}
|
||||
14
meson.build
14
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')
|
||||
|
|
@ -259,16 +259,26 @@ if host_machine.system() == 'windows'
|
|||
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
|
||||
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'
|
||||
|
|
|
|||
|
|
@ -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
|
||||
|
|
@ -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
|
||||
|
|
@ -48,6 +48,7 @@
|
|||
#include <mmsystem.h>
|
||||
#include <process.h>
|
||||
#include <dsound.h>
|
||||
#include <cguid.h>
|
||||
|
||||
namespace {
|
||||
class DirectSoundPlayer2Thread;
|
||||
|
|
@ -319,8 +320,10 @@ void DirectSoundPlayer2Thread::Run()
|
|||
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);
|
||||
|
|
@ -369,7 +372,7 @@ void DirectSoundPlayer2Thread::Run()
|
|||
DWORD buffer_offset = 0;
|
||||
bool playback_should_be_running = false;
|
||||
int current_latency = wanted_latency;
|
||||
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*provider->GetBytesPerSample()/1000;
|
||||
const DWORD wanted_latency_bytes = wanted_latency*waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t)/1000;
|
||||
|
||||
while (running)
|
||||
{
|
||||
|
|
@ -422,7 +425,7 @@ void DirectSoundPlayer2Thread::Run()
|
|||
if (bytes_filled < wanted_latency_bytes)
|
||||
{
|
||||
// Very short playback length, do without streaming playback
|
||||
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
|
||||
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
|
||||
if (FAILED(bfr->Play(0, 0, 0)))
|
||||
REPORT_ERROR("Could not start single-buffer playback.")
|
||||
}
|
||||
|
|
@ -553,7 +556,7 @@ do_fill_buffer:
|
|||
else if (bytes_filled < wanted_latency_bytes)
|
||||
{
|
||||
// Didn't fill as much as we wanted to, let's get back to filling sooner than normal
|
||||
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*provider->GetBytesPerSample());
|
||||
current_latency = (bytes_filled*1000) / (waveFormat.nSamplesPerSec*/*provider->GetBytesPerSample()*/sizeof(int16_t));
|
||||
}
|
||||
else
|
||||
{
|
||||
|
|
@ -577,7 +580,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
|
|||
{
|
||||
// Assume buffers have been locked and are ready to be filled
|
||||
|
||||
DWORD bytes_per_frame = provider->GetChannels() * provider->GetBytesPerSample();
|
||||
DWORD bytes_per_frame = /*provider->GetChannels() * provider->GetBytesPerSample()*/sizeof(int16_t);
|
||||
DWORD buf1szf = buf1sz / bytes_per_frame;
|
||||
DWORD buf2szf = buf2sz / bytes_per_frame;
|
||||
|
||||
|
|
@ -608,7 +611,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
|
|||
buf2sz = 0;
|
||||
}
|
||||
|
||||
provider->GetAudioWithVolume(buf1, input_frame, buf1szf, volume);
|
||||
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf1), input_frame, buf1szf, volume);
|
||||
|
||||
input_frame += buf1szf;
|
||||
}
|
||||
|
|
@ -621,7 +624,7 @@ DWORD DirectSoundPlayer2Thread::FillAndUnlockBuffers(void *buf1, DWORD buf1sz, v
|
|||
buf2sz = buf2szf * bytes_per_frame;
|
||||
}
|
||||
|
||||
provider->GetAudioWithVolume(buf2, input_frame, buf2szf, volume);
|
||||
provider->GetInt16MonoAudioWithVolume(reinterpret_cast<int16_t*>(buf2), input_frame, buf2szf, volume);
|
||||
|
||||
input_frame += buf2szf;
|
||||
}
|
||||
|
|
@ -932,4 +935,4 @@ std::unique_ptr<AudioPlayer> CreateDirectSound2Player(agi::AudioProvider *provid
|
|||
return agi::make_unique<DirectSoundPlayer2>(provider, parent);
|
||||
}
|
||||
|
||||
#endif // WITH_DIRECTSOUND
|
||||
#endif // WITH_DIRECTSOUND
|
||||
|
|
@ -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
|
||||
|
|
@ -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
|
||||
|
|
@ -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
|
||||
|
|
@ -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
|
||||
|
|
@ -236,7 +236,7 @@ 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]);
|
||||
}
|
||||
|
|
@ -551,4 +551,4 @@ TEST(lagi_audio, wave64_truncated) {
|
|||
}
|
||||
|
||||
agi::fs::Remove(path);
|
||||
}
|
||||
}
|
||||
Loading…
Reference in New Issue