Sync from bytedesk-private: update

modules/python/vendors/FunASR/runtime/onnxruntime/include/audio.h

@@ -0,0 +1,116 @@
+#ifndef AUDIO_H
+#define AUDIO_H
+
+#include <queue>
+#include <stdint.h>
+#include "vad-model.h"
+#include "offline-stream.h"
+#include "com-define.h"
+
+#ifndef WAV_HEADER_SIZE
+#define WAV_HEADER_SIZE 44
+#endif
+
+using namespace std;
+namespace funasr {
+
+class AudioFrame {
+  private:
+    int start;
+    int end;
+
+
+  public:
+    AudioFrame();
+    AudioFrame(int len);
+    AudioFrame(const AudioFrame &other);
+    AudioFrame(int start, int end, bool is_final);
+
+    ~AudioFrame();
+    int SetStart(int val);
+    int SetEnd(int val);
+    int GetStart();
+    int GetLen();
+    int Disp();
+    // 2pass
+    bool is_final = false;
+    float* data = nullptr;
+    int len;
+    int global_start = 0; // the start of a frame in the global time axis. in ms
+    int global_end = 0;   // the end of a frame in the global time axis. in ms
+};
+
+#ifdef _WIN32
+#ifdef _FUNASR_API_EXPORT
+#define DLLAPI __declspec(dllexport)
+#else
+#define DLLAPI __declspec(dllimport)
+#endif
+#else
+#define DLLAPI 
+#endif
+class DLLAPI Audio {
+  private:
+    float *speech_data=nullptr;
+    int16_t *speech_buff=nullptr;
+    char* speech_char=nullptr;
+    int speech_len;
+    int speech_align_len;
+    float align_size;
+    int data_type;
+    queue<AudioFrame *> frame_queue;
+    queue<AudioFrame *> asr_online_queue;
+    queue<AudioFrame *> asr_offline_queue;
+    int dest_sample_rate;
+  public:
+    Audio(int data_type);
+    Audio(int model_sample_rate,int data_type);
+    Audio(int model_sample_rate,int data_type, int size);
+    ~Audio();
+    void ClearQueue(std::queue<AudioFrame*>& q);
+    void Disp();
+    void WavResample(int32_t sampling_rate, const float *waveform, int32_t n);
+    bool LoadWav(const char* buf, int n_len, int32_t* sampling_rate);
+    bool LoadWav(const char* filename, int32_t* sampling_rate, bool resample=true);
+    bool LoadWav2Char(const char* filename, int32_t* sampling_rate);
+    bool LoadPcmwav(const char* buf, int n_file_len, int32_t* sampling_rate);
+    bool LoadPcmwav(const char* filename, int32_t* sampling_rate, bool resample=true);
+    bool LoadPcmwav2Char(const char* filename, int32_t* sampling_rate);
+    bool LoadOthers2Char(const char* filename);
+    bool FfmpegLoad(const char *filename, bool copy2char=false);
+    bool FfmpegLoad(const char* buf, int n_file_len);
+    int FetchChunck(AudioFrame *&frame);
+    int FetchTpass(AudioFrame *&frame);
+    int Fetch(float *&dout, int &len, int &flag);
+    int Fetch(float *&dout, int &len, int &flag, float &start_time);
+    int Fetch(float **&dout, int *&len, int *&flag, float*& start_time, int batch_size, int &batch_in);
+    int FetchDynamic(float **&dout, int *&len, int *&flag, float*& start_time, int batch_size, int &batch_in);
+    void Padding();
+    void Split(OfflineStream* offline_streamj);
+    void CutSplit(OfflineStream* offline_streamj, std::vector<int> &index_vector);
+    void Split(VadModel* vad_obj, vector<std::vector<int>>& vad_segments, bool input_finished=true);
+    void Split(VadModel* vad_obj, int chunk_len, bool input_finished=true, ASR_TYPE asr_mode=ASR_TWO_PASS);
+    float GetTimeLen();
+    int GetQueueSize() { return (int)frame_queue.size(); }
+    char* GetSpeechChar(){return speech_char;}
+    int GetSpeechLen(){return speech_len;}
+
+    // 2pass
+    vector<float> all_samples;
+    int offset = 0;
+    int speech_start=-1, speech_end=0;
+    int speech_offline_start=-1;
+
+    int seg_sample = MODEL_SAMPLE_RATE/1000;
+    bool LoadPcmwavOnline(const char* buf, int n_file_len, int32_t* sampling_rate);
+    void ResetIndex(){
+      speech_start=-1;
+      speech_end=0;
+      speech_offline_start=-1;
+      offset = 0;
+      all_samples.clear();
+    }
+};
+
+} // namespace funasr
+#endif