#pragma once #include "util/types.hpp" #include "Utilities/mutex.h" #include "Utilities/StrFmt.h" #include enum : u32 { DEFAULT_AUDIO_SAMPLING_RATE = 48000, MAX_AUDIO_BUFFERS = 64, AUDIO_BUFFER_SAMPLES = 256, AUDIO_MAX_CHANNELS = 8, }; enum class AudioFreq : u32 { FREQ_32K = 32000, FREQ_44K = 44100, FREQ_48K = 48000, FREQ_88K = 88200, FREQ_96K = 96000, FREQ_176K = 176400, FREQ_192K = 192000, }; enum class AudioSampleSize : u32 { FLOAT = sizeof(float), S16 = sizeof(s16), }; enum class AudioChannelCnt : u32 { STEREO = 2, SURROUND_5_1 = 6, SURROUND_7_1 = 8, }; class AudioBackend { public: struct VolumeParam { f32 initial_volume = 1.0f; f32 current_volume = 1.0f; f32 target_volume = 1.0f; u32 freq = 48000; u32 ch_cnt = 2; }; AudioBackend(); virtual ~AudioBackend() = default; /* * Virtual methods */ virtual std::string_view GetName() const = 0; // (Re)create output stream with new parameters. Blocks until data callback returns. // Should return 'true' on success. virtual bool Open(AudioFreq freq, AudioSampleSize sample_size, AudioChannelCnt ch_cnt) = 0; // Reset backend state. Blocks until data callback returns. virtual void Close() = 0; // Sets write callback. It's called when backend requests new data to be sent. // Callback should return number of submitted bytes. Calling other backend functions from callback is unsafe. virtual void SetWriteCallback(std::function cb) = 0; // Sets error callback. It's called when backend detects uncorrectable error condition in audio chain. // Calling other backend functions from callback is unsafe. virtual void SetErrorCallback(std::function cb); /* * All functions below require that Open() was called prior. */ // Returns length of one write callback frame in seconds. Open() must be called prior. virtual f64 GetCallbackFrameLen() = 0; // Returns true if audio is currently being played, false otherwise. Reflects end result of Play() and Pause() calls. virtual bool IsPlaying() { return m_playing; } // Start playing enqueued data. virtual void Play() = 0; // Pause playing enqueued data. No additional callbacks will be issued. Blocks until data callback returns. virtual void Pause() = 0; /* * This virtual method should be reimplemented if backend can fail to be initialized under non-error conditions * eg. when there is no audio devices attached */ virtual bool Initialized() { return true; } /* * This virtual method should be reimplemented if backend can fail during normal operation */ virtual bool Operational() { return true; } /* * Helper methods */ u32 get_sampling_rate() const; u32 get_sample_size() const; u32 get_channels() const; bool get_convert_to_s16() const; /* * Convert float buffer to s16 one. src and dst could be the same. cnt is number of buffer elements. */ static void convert_to_s16(u32 cnt, const f32* src, void* dst); /* * Apply volume parameters to the buffer. Gradually changes volume. src and dst could be the same. * Number of channels must be >1 and multiple of 2. * sample_cnt is number of buffer elements. Returns current volume. */ static f32 apply_volume(const VolumeParam& param, u32 sample_cnt, const f32* src, f32* dst); /* * Apply volume value to the buffer. src and dst could be the same. sample_cnt is number of buffer elements. * Returns current volume. */ static void apply_volume_static(f32 vol, u32 sample_cnt, const f32* src, f32* dst); /* * Downmix audio stream. */ template static void downmix(u32 sample_cnt, const f32* src, f32* dst) { static_assert(from == AudioChannelCnt::SURROUND_5_1 || from == AudioChannelCnt::SURROUND_7_1, "Cannot downmix FROM channel count"); static_assert(static_cast(from) > static_cast(to), "FROM channel count must be bigger than TO"); static constexpr f32 center_coef = std::numbers::sqrt2_v / 2; static constexpr f32 surround_coef = std::numbers::sqrt2_v / 2; for (u32 src_sample = 0, dst_sample = 0; src_sample < sample_cnt; src_sample += static_cast(from), dst_sample += static_cast(to)) { const f32 left = src[src_sample + 0]; const f32 right = src[src_sample + 1]; const f32 center = src[src_sample + 2]; const f32 low_freq = src[src_sample + 3]; if constexpr (from == AudioChannelCnt::SURROUND_5_1) { static_assert(to == AudioChannelCnt::STEREO, "Invalid TO channel count"); const f32 side_left = src[src_sample + 4]; const f32 side_right = src[src_sample + 5]; const f32 mid = center * center_coef; dst[dst_sample + 0] = left + mid + side_left * surround_coef; dst[dst_sample + 1] = right + mid + side_right * surround_coef; } else if constexpr (from == AudioChannelCnt::SURROUND_7_1) { static_assert(to == AudioChannelCnt::STEREO || to == AudioChannelCnt::SURROUND_5_1, "Invalid TO channel count"); const f32 rear_left = src[src_sample + 4]; const f32 rear_right = src[src_sample + 5]; const f32 side_left = src[src_sample + 6]; const f32 side_right = src[src_sample + 7]; if constexpr (to == AudioChannelCnt::SURROUND_5_1) { dst[dst_sample + 0] = left; dst[dst_sample + 1] = right; dst[dst_sample + 2] = center; dst[dst_sample + 3] = low_freq; dst[dst_sample + 4] = side_left + rear_left; dst[dst_sample + 5] = side_right + rear_right; } else { const f32 mid = center * center_coef; dst[dst_sample + 0] = left + mid + (side_left + rear_left) * surround_coef; dst[dst_sample + 1] = right + mid + (side_right + rear_right) * surround_coef; } } } } /* * Normalize float samples in range from -1.0 to 1.0. */ static void normalize(u32 sample_cnt, const f32* src, f32* dst); protected: AudioSampleSize m_sample_size = AudioSampleSize::FLOAT; AudioFreq m_sampling_rate = AudioFreq::FREQ_48K; AudioChannelCnt m_channels = AudioChannelCnt::STEREO; shared_mutex m_error_cb_mutex{}; std::function m_error_callback{}; bool m_playing = false; private: static constexpr f32 VOLUME_CHANGE_DURATION = 0.016f; // sec };