自定 AudioStream

簡介

AudioStream 是所有音訊發射物件的基礎類別。AudioStreamPlayer 繫結到 AudioStream 上，以將 PCM 資料發射至 AudioServer 內。AudioServer 則負責管理音訊驅動程式。

所有的音訊資源都需要兩個音訊基礎類別：AudioStream 與 AudioStreamPlayback。作為資料容器，AudioStream 包含了音訊資源，並會將其自身暴露給 GDScript。AudioStream 也會將其自身參照到內部的自定 AudioStreamPlayback，用來將 AudioStream 翻譯至 PCM 資料。

本篇指南假設讀者已瞭解如何建立 C++ 模組。否則，請先參考這篇指南： 以 C++ 語言自定模組 。

參考資料：

• servers/audio/audio_stream.h

• scene/audio/audioplayer.cpp

可以做什麼？

• 繫結外部函式庫 (如 Wwise, FMOD…等)。

• 新增自定音訊佇列

• 新增更多音訊格式的支援

建立 AudioStream

AudioStream 由三個元件組成：資料容器、串流名稱以及 AudioStreamPlayback 友類產生器 (Friend Class Generator)。音訊資料可以通過數種方式載入，如用於音調產生器的內部計數器、內／外部緩衝區或檔案參照。

有的 AudioStream 必須要是無狀態的，如由 ResourceLoader 載入的物件。ResourceLoader 會載入一次，然後無論對特定資源呼叫多少次 load ，都會參照到相同的物件上。因此，播放狀態必須要在 AudioStreamPlayback 內自封閉。

/* audiostream_mytone.h */

#include "core/reference.h"
#include "core/resource.h"
#include "servers/audio/audio_stream.h"

class AudioStreamMyTone : public AudioStream {
        GDCLASS(AudioStreamMyTone, AudioStream)

private:
        friend class AudioStreamPlaybackMyTone;
        uint64_t pos;
        int mix_rate;
        bool stereo;
        int hz;

public:
        void reset();
        void set_position(uint64_t pos);
        virtual Ref<AudioStreamPlayback> instance_playback();
        virtual String get_stream_name() const;
        void gen_tone(int16_t *pcm_buf, int size);
        virtual float get_length() const { return 0; } // if supported, otherwise return 0
        AudioStreamMyTone();

protected:
        static void _bind_methods();
};

/* audiostream_mytone.cpp */

#include "audiostream_mytone.h"

AudioStreamMyTone::AudioStreamMyTone()
                : mix_rate(44100), stereo(false), hz(639) {
}

Ref<AudioStreamPlayback> AudioStreamMyTone::instance_playback() {
        Ref<AudioStreamPlaybackMyTone> talking_tree;
        talking_tree.instance();
        talking_tree->base = Ref<AudioStreamMyTone>(this);
        return talking_tree;
}

String AudioStreamMyTone::get_stream_name() const {
        return "MyTone";
}
void AudioStreamMyTone::reset() {
        set_position(0);
}
void AudioStreamMyTone::set_position(uint64_t p) {
        pos = p;
}
void AudioStreamMyTone::gen_tone(int16_t *pcm_buf, int size) {
        for (int i = 0; i < size; i++) {
                pcm_buf[i] = 32767.0 * sin(2.0 * Math_PI * double(pos + i) / (double(mix_rate) / double(hz)));
        }
        pos += size;
}
void AudioStreamMyTone::_bind_methods() {
        ClassDB::bind_method(D_METHOD("reset"), &AudioStreamMyTone::reset);
        ClassDB::bind_method(D_METHOD("get_stream_name"), &AudioStreamMyTone::get_stream_name);
}

參考資料：

• servers/audio/audio_stream.h

建立 AudioStreamPlayback

AudioStreamPlayer 使用 mix 回呼來取得 PCM 資料。該回呼必須要符合採樣率，並填充緩衝區。

由於 AudioStreamPlayback 是由音訊執行緒控制的，因此不可進行 I/O 與動態記憶體分配。

/*  audiostreamplayer_mytone.h */

#include "core/reference.h"
#include "core/resource.h"
#include "servers/audio/audio_stream.h"

class AudioStreamPlaybackMyTone : public AudioStreamPlayback {
        GDCLASS(AudioStreamPlaybackMyTone, AudioStreamPlayback)
        friend class AudioStreamMyTone;

private:
        enum {
                PCM_BUFFER_SIZE = 4096
        };
        enum {
                MIX_FRAC_BITS = 13,
                MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
                MIX_FRAC_MASK = MIX_FRAC_LEN - 1,
        };
        void *pcm_buffer;
        Ref<AudioStreamMyTone> base;
        bool active;

public:
        virtual void start(float p_from_pos = 0.0);
        virtual void stop();
        virtual bool is_playing() const;
        virtual int get_loop_count() const; // times it looped
        virtual float get_playback_position() const;
        virtual void seek(float p_time);
        virtual void mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames);
        virtual float get_length() const; // if supported, otherwise return 0
        AudioStreamPlaybackMyTone();
        ~AudioStreamPlaybackMyTone();
};

/* audiostreamplayer_mytone.cpp */

#include "audiostreamplayer_mytone.h"

#include "core/math/math_funcs.h"
#include "core/print_string.h"

AudioStreamPlaybackMyTone::AudioStreamPlaybackMyTone()
                : active(false) {
        AudioServer::get_singleton()->lock();
        pcm_buffer = AudioServer::get_singleton()->audio_data_alloc(PCM_BUFFER_SIZE);
        zeromem(pcm_buffer, PCM_BUFFER_SIZE);
        AudioServer::get_singleton()->unlock();
}
AudioStreamPlaybackMyTone::~AudioStreamPlaybackMyTone() {
        if(pcm_buffer) {
                AudioServer::get_singleton()->audio_data_free(pcm_buffer);
                pcm_buffer = NULL;
        }
}
void AudioStreamPlaybackMyTone::stop() {
        active = false;
        base->reset();
}
void AudioStreamPlaybackMyTone::start(float p_from_pos) {
        seek(p_from_pos);
        active = true;
}
void AudioStreamPlaybackMyTone::seek(float p_time) {
        float max = get_length();
        if (p_time < 0) {
                        p_time = 0;
        }
        base->set_position(uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS);
}
void AudioStreamPlaybackMyTone::mix(AudioFrame *p_buffer, float p_rate, int p_frames) {
        ERR_FAIL_COND(!active);
        if (!active) {
                        return;
        }
        zeromem(pcm_buffer, PCM_BUFFER_SIZE);
        int16_t *buf = (int16_t *)pcm_buffer;
        base->gen_tone(buf, p_frames);

        for(int i = 0; i < p_frames; i++) {
                float sample = float(buf[i]) / 32767.0;
                p_buffer[i] = AudioFrame(sample, sample);
        }
}
int AudioStreamPlaybackMyTone::get_loop_count() const {
        return 0;
}
float AudioStreamPlaybackMyTone::get_playback_position() const {
        return 0.0;
}
float AudioStreamPlaybackMyTone::get_length() const {
        return 0.0;
}
bool AudioStreamPlaybackMyTone::is_playing() const {
        return active;
}

重新採樣

Godoto 的 AudioServer 目前使用 44100 Hz 的採樣率。當需要其他如 48000 的採樣率時，則必須要主動提供採樣率，或是使用 AudioStreamPlaybackResampled。Godot 提供了三次方內插補點 (Cubic Interpolation) 用於音訊重新採樣。

AudioStreamPlaybackResampled 不是複寫 mix ，而是使用 _mix_internal 來查詢 AudioFrame，並使用 get_stream_sampling_rate 來查詢目前的混合率。

#include "core/reference.h"
#include "core/resource.h"
#include "servers/audio/audio_stream.h"

class AudioStreamMyToneResampled;

class AudioStreamPlaybackResampledMyTone : public AudioStreamPlaybackResampled {
        GDCLASS(AudioStreamPlaybackResampledMyTone, AudioStreamPlaybackResampled)
        friend class AudioStreamMyToneResampled;

private:
        enum {
                PCM_BUFFER_SIZE = 4096
        };
        enum {
                MIX_FRAC_BITS = 13,
                MIX_FRAC_LEN = (1 << MIX_FRAC_BITS),
                MIX_FRAC_MASK = MIX_FRAC_LEN - 1,
        };
        void *pcm_buffer;
        Ref<AudioStreamMyToneResampled> base;
        bool active;

protected:
        virtual void _mix_internal(AudioFrame *p_buffer, int p_frames);

public:
        virtual void start(float p_from_pos = 0.0);
        virtual void stop();
        virtual bool is_playing() const;
        virtual int get_loop_count() const; // times it looped
        virtual float get_playback_position() const;
        virtual void seek(float p_time);
        virtual float get_length() const; // if supported, otherwise return 0
        virtual float get_stream_sampling_rate();
        AudioStreamPlaybackResampledMyTone();
        ~AudioStreamPlaybackResampledMyTone();
};

#include "mytone_audiostream_resampled.h"

#include "core/math/math_funcs.h"
#include "core/print_string.h"

AudioStreamPlaybackResampledMyTone::AudioStreamPlaybackResampledMyTone()
                : active(false) {
        AudioServer::get_singleton()->lock();
        pcm_buffer = AudioServer::get_singleton()->audio_data_alloc(PCM_BUFFER_SIZE);
        zeromem(pcm_buffer, PCM_BUFFER_SIZE);
        AudioServer::get_singleton()->unlock();
}
AudioStreamPlaybackResampledMyTone::~AudioStreamPlaybackResampledMyTone() {
        if (pcm_buffer) {
                AudioServer::get_singleton()->audio_data_free(pcm_buffer);
                pcm_buffer = NULL;
        }
}
void AudioStreamPlaybackResampledMyTone::stop() {
        active = false;
        base->reset();
}
void AudioStreamPlaybackResampledMyTone::start(float p_from_pos) {
        seek(p_from_pos);
        active = true;
}
void AudioStreamPlaybackResampledMyTone::seek(float p_time) {
        float max = get_length();
        if (p_time < 0) {
                        p_time = 0;
        }
        base->set_position(uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS);
}
void AudioStreamPlaybackResampledMyTone::_mix_internal(AudioFrame *p_buffer, int p_frames) {
        ERR_FAIL_COND(!active);
        if (!active) {
                return;
        }
        zeromem(pcm_buffer, PCM_BUFFER_SIZE);
        int16_t *buf = (int16_t *)pcm_buffer;
        base->gen_tone(buf, p_frames);

        for(int i = 0;  i < p_frames; i++) {
                float sample = float(buf[i]) / 32767.0;
                        p_buffer[i] = AudioFrame(sample, sample);
        }
}
float AudioStreamPlaybackResampledMyTone::get_stream_sampling_rate() {
        return float(base->mix_rate);
}
int AudioStreamPlaybackResampledMyTone::get_loop_count() const {
        return 0;
}
float AudioStreamPlaybackResampledMyTone::get_playback_position() const {
        return 0.0;
}
float AudioStreamPlaybackResampledMyTone::get_length() const {
        return 0.0;
}
bool AudioStreamPlaybackResampledMyTone::is_playing() const {
        return active;
}

參考資料：

• core/math/audio_frame.h

• servers/audio/audio_stream.h

• scene/audio/audioplayer.cpp