AudioStream: Re-add SDL backend
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9703542775
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@ -365,6 +365,11 @@ std::unique_ptr<AudioStream> Host::CreateAudioStream(AudioBackend backend, u32 s
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return AudioStream::CreateCubebAudioStream(sample_rate, channels, buffer_ms, latency_ms, stretch);
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#endif
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#ifdef ENABLE_SDL2
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case AudioBackend::SDL:
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return AudioStream::CreateSDLAudioStream(sample_rate, channels, buffer_ms, latency_ms, stretch);
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#endif
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#ifdef _WIN32
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case AudioBackend::XAudio2:
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return AudioStream::CreateXAudio2Stream(sample_rate, channels, buffer_ms, latency_ms, stretch);
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@ -1559,6 +1559,9 @@ static constexpr const std::array s_audio_backend_names = {
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#ifdef ENABLE_CUBEB
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"Cubeb",
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#endif
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#ifdef ENABLE_SDL2
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"SDL",
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#endif
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#ifdef _WIN32
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"XAudio2",
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#endif
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@ -1571,6 +1574,9 @@ static constexpr const std::array s_audio_backend_display_names = {
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#ifdef ENABLE_CUBEB
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TRANSLATE_NOOP("AudioBackend", "Cubeb"),
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#endif
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#ifdef ENABLE_SDL2
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TRANSLATE_NOOP("AudioBackend", "SDL"),
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#endif
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#ifdef _WIN32
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TRANSLATE_NOOP("AudioBackend", "XAudio2"),
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#endif
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@ -484,6 +484,8 @@ struct Settings
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static constexpr AudioBackend DEFAULT_AUDIO_BACKEND = AudioBackend::XAudio2;
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#elif defined(__ANDROID__)
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static constexpr AudioBackend DEFAULT_AUDIO_BACKEND = AudioBackend::AAudio;
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#elif defined(ENABLE_SDL2)
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static constexpr AudioBackend DEFAULT_AUDIO_BACKEND = AudioBackend::SDL;
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#else
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static constexpr AudioBackend DEFAULT_AUDIO_BACKEND = AudioBackend::Null;
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#endif
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@ -190,6 +190,9 @@ enum class AudioBackend : u8
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#ifdef ENABLE_CUBEB
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Cubeb,
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#endif
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#ifdef ENABLE_SDL2
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SDL,
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#endif
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#ifdef _WIN32
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XAudio2,
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#endif
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@ -190,6 +190,7 @@ endif()
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if(ENABLE_SDL2)
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target_sources(util PRIVATE
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sdl_audio_stream.cpp
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sdl_input_source.cpp
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sdl_input_source.h
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)
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@ -89,10 +89,10 @@ u32 AudioStream::GetBufferedFramesRelaxed() const
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return (wpos + m_buffer_size - rpos) % m_buffer_size;
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}
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void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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void AudioStream::ReadFrames(s16* samples, u32 num_frames)
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{
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const u32 available_frames = GetBufferedFramesRelaxed();
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u32 frames_to_read = nFrames;
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u32 frames_to_read = num_frames;
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u32 silence_frames = 0;
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if (m_filling)
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@ -102,7 +102,7 @@ void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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if (available_frames < toFill)
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{
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silence_frames = nFrames;
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silence_frames = num_frames;
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frames_to_read = 0;
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}
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else
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@ -133,7 +133,7 @@ void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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// towards the end of the buffer
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if (end > 0)
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{
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std::memcpy(bData, &m_buffer[rpos], sizeof(s32) * end);
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std::memcpy(samples, &m_buffer[rpos], sizeof(s32) * end);
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rpos += end;
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rpos = (rpos == m_buffer_size) ? 0 : rpos;
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}
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@ -142,7 +142,7 @@ void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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const u32 start = frames_to_read - end;
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if (start > 0)
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{
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std::memcpy(&bData[end * 2], &m_buffer[0], sizeof(s32) * start);
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std::memcpy(&samples[end * 2], &m_buffer[0], sizeof(s32) * start);
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rpos = start;
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}
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@ -156,15 +156,15 @@ void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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// super basic resampler - spread the input samples evenly across the output samples. will sound like ass and have
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// aliasing, but better than popping by inserting silence.
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const u32 increment =
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static_cast<u32>(65536.0f * (static_cast<float>(frames_to_read / m_channels) / static_cast<float>(nFrames)));
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static_cast<u32>(65536.0f * (static_cast<float>(frames_to_read / m_channels) / static_cast<float>(num_frames)));
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s16* resample_ptr = static_cast<s16*>(alloca(sizeof(s16) * frames_to_read));
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std::memcpy(resample_ptr, bData, sizeof(s16) * frames_to_read);
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std::memcpy(resample_ptr, samples, sizeof(s16) * frames_to_read);
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s16* out_ptr = bData;
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s16* out_ptr = samples;
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const u32 copy_stride = sizeof(SampleType) * m_channels;
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u32 resample_subpos = 0;
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for (u32 i = 0; i < nFrames; i++)
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for (u32 i = 0; i < num_frames; i++)
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{
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std::memcpy(out_ptr, resample_ptr, copy_stride);
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out_ptr += m_channels;
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@ -174,16 +174,28 @@ void AudioStream::ReadFrames(s16* bData, u32 nFrames)
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resample_subpos %= 65536u;
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}
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Log_VerbosePrintf("Audio buffer underflow, resampled %u frames to %u", frames_to_read, nFrames);
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Log_VerbosePrintf("Audio buffer underflow, resampled %u frames to %u", frames_to_read, num_frames);
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}
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else
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{
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// no data, fall back to silence
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std::memset(bData + frames_to_read, 0, sizeof(s32) * silence_frames);
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std::memset(samples + frames_to_read, 0, sizeof(s32) * silence_frames);
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}
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}
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}
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void AudioStream::ApplyVolume(s16* samples, u32 num_frames)
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{
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if (m_volume == 100)
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return;
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const s32 volume_mult = static_cast<s32>(m_volume) * 32768;
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const u32 num_samples = num_frames * m_channels;
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while (num_samples > 0)
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*samples = static_cast<s16>((static_cast<s16>(*samples) * volume_mult) >> 15);
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}
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void AudioStream::InternalWriteFrames(s32* bData, u32 nSamples)
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{
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const u32 free = m_buffer_size - GetBufferedFramesRelaxed();
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@ -85,6 +85,10 @@ public:
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static std::vector<std::string> GetCubebDriverNames();
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static std::vector<std::pair<std::string, std::string>> GetCubebOutputDevices(const char* driver);
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#endif
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#ifdef ENABLE_SDL2
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static std::unique_ptr<AudioStream> CreateSDLAudioStream(u32 sample_rate, u32 channels, u32 buffer_ms, u32 latency_ms,
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AudioStretchMode stretch);
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#endif
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#ifdef _WIN32
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static std::unique_ptr<AudioStream> CreateXAudio2Stream(u32 sample_rate, u32 channels, u32 buffer_ms, u32 latency_ms,
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AudioStretchMode stretch);
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@ -94,7 +98,8 @@ protected:
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AudioStream(u32 sample_rate, u32 channels, u32 buffer_ms, AudioStretchMode stretch);
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void BaseInitialize();
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void ReadFrames(s16* bData, u32 nSamples);
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void ReadFrames(s16* samples, u32 num_frames);
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void ApplyVolume(s16* samples, u32 num_frames);
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u32 m_sample_rate = 0;
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u32 m_channels = 0;
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@ -0,0 +1,136 @@
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// SPDX-FileCopyrightText: 2019-2024 Connor McLaughlin <stenzek@gmail.com>
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// SPDX-License-Identifier: (GPL-3.0 OR CC-BY-NC-ND-4.0)
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#include "audio_stream.h"
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#include "common/assert.h"
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#include "common/log.h"
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#include <SDL.h>
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Log_SetChannel(SDLAudioStream);
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namespace {
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class SDLAudioStream final : public AudioStream
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{
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public:
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SDLAudioStream(u32 sample_rate, u32 channels, u32 buffer_ms, AudioStretchMode stretch);
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~SDLAudioStream();
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void SetPaused(bool paused) override;
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void SetOutputVolume(u32 volume) override;
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bool OpenDevice(u32 latency_ms);
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void CloseDevice();
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protected:
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ALWAYS_INLINE bool IsOpen() const { return (m_device_id != 0); }
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static void AudioCallback(void* userdata, uint8_t* stream, int len);
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u32 m_device_id = 0;
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};
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} // namespace
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static bool InitializeSDLAudio()
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{
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static bool initialized = false;
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if (initialized)
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return true;
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// May as well keep it alive until the process exits.
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const int error = SDL_InitSubSystem(SDL_INIT_AUDIO);
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if (error != 0)
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{
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Log_ErrorFmt("SDL_InitSubSystem(SDL_INIT_AUDIO) returned {}", error);
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return false;
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}
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std::atexit([]() { SDL_QuitSubSystem(SDL_INIT_AUDIO); });
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initialized = true;
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return true;
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}
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SDLAudioStream::SDLAudioStream(u32 sample_rate, u32 channels, u32 buffer_ms, AudioStretchMode stretch)
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: AudioStream(sample_rate, channels, buffer_ms, stretch)
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{
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}
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SDLAudioStream::~SDLAudioStream()
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{
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if (IsOpen())
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SDLAudioStream::CloseDevice();
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}
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std::unique_ptr<AudioStream> AudioStream::CreateSDLAudioStream(u32 sample_rate, u32 channels, u32 buffer_ms,
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u32 latency_ms, AudioStretchMode stretch)
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{
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if (!InitializeSDLAudio())
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return {};
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std::unique_ptr<SDLAudioStream> stream = std::make_unique<SDLAudioStream>(sample_rate, channels, buffer_ms, stretch);
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if (!stream->OpenDevice(latency_ms))
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stream.reset();
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return stream;
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}
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bool SDLAudioStream::OpenDevice(u32 latency_ms)
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{
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DebugAssert(!IsOpen());
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SDL_AudioSpec spec = {};
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spec.freq = m_sample_rate;
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spec.channels = static_cast<Uint8>(m_channels);
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spec.format = AUDIO_S16;
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spec.samples = static_cast<Uint16>(GetBufferSizeForMS(m_sample_rate, (latency_ms == 0) ? m_buffer_ms : latency_ms));
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spec.callback = AudioCallback;
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spec.userdata = static_cast<void*>(this);
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SDL_AudioSpec obtained_spec = {};
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m_device_id = SDL_OpenAudioDevice(nullptr, 0, &spec, &obtained_spec, SDL_AUDIO_ALLOW_SAMPLES_CHANGE);
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if (m_device_id == 0)
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{
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Log_ErrorFmt("SDL_OpenAudioDevice() failed: {}", SDL_GetError());
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return false;
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}
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Log_DevFmt("Requested {} frame buffer, got {} frame buffer", spec.samples, obtained_spec.samples);
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BaseInitialize();
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m_volume = 100;
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m_paused = false;
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SDL_PauseAudioDevice(m_device_id, 0);
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return true;
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}
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void SDLAudioStream::SetPaused(bool paused)
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{
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if (m_paused == paused)
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return;
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SDL_PauseAudioDevice(m_device_id, paused ? 1 : 0);
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m_paused = paused;
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}
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void SDLAudioStream::CloseDevice()
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{
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SDL_CloseAudioDevice(m_device_id);
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m_device_id = 0;
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}
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void SDLAudioStream::AudioCallback(void* userdata, uint8_t* stream, int len)
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{
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SDLAudioStream* const this_ptr = static_cast<SDLAudioStream*>(userdata);
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const u32 num_frames = len / sizeof(SampleType) / this_ptr->m_channels;
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this_ptr->ReadFrames(reinterpret_cast<SampleType*>(stream), num_frames);
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this_ptr->ApplyVolume(reinterpret_cast<SampleType*>(stream), num_frames);
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}
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void SDLAudioStream::SetOutputVolume(u32 volume)
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{
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m_volume = volume;
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}
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@ -203,6 +203,7 @@
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<ClCompile Include="postprocessing_shader.cpp" />
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<ClCompile Include="postprocessing_shader_fx.cpp" />
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<ClCompile Include="postprocessing_shader_glsl.cpp" />
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<ClCompile Include="sdl_audio_stream.cpp" />
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<ClCompile Include="sdl_input_source.cpp" />
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<ClCompile Include="shadergen.cpp" />
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<ClCompile Include="shiftjis.cpp" />
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@ -156,6 +156,7 @@
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<ClCompile Include="opengl_context_egl_x11.cpp" />
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<ClCompile Include="opengl_context_wgl.cpp" />
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<ClCompile Include="image.cpp" />
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<ClCompile Include="sdl_audio_stream.cpp" />
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</ItemGroup>
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<ItemGroup>
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<None Include="metal_shaders.metal" />
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