自訂 AudioStream
前言
AudioStream 是所有音訊發出物件的基底類別。AudioStreamPlayer 會繫結到 AudioStream 上,將 PCM 資料傳送至 AudioServer,由 AudioServer 管理音訊驅動程式。
所有音訊資源都需有兩個基礎類別:AudioStream 以及 AudioStreamPlayback。AudioStream 作為資料容器,會包含音訊資源並對 GDScript 開放存取。AudioStream 會參照其內部自訂的 AudioStreamPlayback,將 AudioStream 轉換成 PCM 資料。
本指南假設讀者已熟悉如何建立 C++ 模組。如尚未了解,請先參考這篇指南:以 C++ 自訂模組。
參考資料:
用途說明?
繫結外部函式庫(如 Wwise、FMOD 等)。
新增自訂音訊佇列
增加更多音訊格式的支援
建立 AudioStream
一個 AudioStream 由三個組成元件構成:資料容器、串流名稱,以及 AudioStreamPlayback 友類產生器。音訊資料可透過多種方式載入,例如作為音調產生器的內部計數器、內部或外部緩衝區,或是檔案參照。
有些 AudioStream 需為無狀態(stateless),如由 ResourceLoader 載入的物件。ResourceLoader 只會載入一次,無論針對特定資源呼叫幾次 load,都會參照同一個物件。因此,播放狀態必須完全封裝於 AudioStreamPlayback 之中。
#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();
};
#include "audiostream_mytone.h"
AudioStreamMyTone::AudioStreamMyTone()
: mix_rate(44100), stereo(false), hz(639) {
}
Ref<AudioStreamPlayback> AudioStreamMyTone::instance_playback() {
Ref<AudioStreamPlaybackMyTone> talking_tree;
talking_tree.instantiate();
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:
建立 AudioStreamPlayback
AudioStreamPlayer 會使用 mix 回呼來取得 PCM 資料。此回呼必須符合取樣率,並填滿緩衝區。
由於 AudioStreamPlayback 受音訊執行緒控制,因此禁止進行 I/O 或動態記憶體分配。
#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();
};
#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;
}
重新取樣
Godot 的 AudioServer 目前採用 44100 Hz 取樣率。若需其他取樣率(如 48000 Hz),可自行提供或使用 AudioStreamPlaybackResampled。Godot 提供三次方內插(cubic interpolation)進行音訊重新取樣。
AudioStreamPlaybackResampled 並非複寫 mix,而是使用 _mix_internal 查詢 AudioFrame,並以 get_stream_sampling_rate 取得目前混音取樣率。
#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;
}