libaudions.cpp 7.2 KB

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  1. // libaudions.cpp : Defines the exported functions for the DLL application.
  2. //
  3. #include "stdafx.h"
  4. #include "libaudions.h"
  5. #include "signal_processing_library.h"
  6. //默认是10ms的帧长度
  7. #ifndef RVC_NS_AUDIO_FRAME_TIME
  8. #define RVC_NS_AUDIO_FRAME_TIME 10
  9. #endif
  10. AudioNsImpl::AudioNsImpl(audions_callback_t* pCallback)
  11. {
  12. memcpy(&m_nscallback, pCallback, sizeof(audions_callback_t));
  13. NsLog("AudioNsImpl construction.");
  14. m_NsHandle = NULL;
  15. if (0 == WebRtcNs_Create(&m_NsHandle)){
  16. NsLog("AudioNsImpl success.");
  17. }
  18. m_audio_sample_rate = 16000;
  19. m_audio_capture_peroid = 20;
  20. m_ns_policy = 2;
  21. }
  22. AudioNsImpl::~AudioNsImpl()
  23. {
  24. NsLog("AudioNsImpl Deconstruction.");
  25. if (NULL != m_NsHandle){
  26. int iRet = WebRtcNs_Free(m_NsHandle);
  27. if (0 == iRet){
  28. NsLog("WebRtcNs Free success.");
  29. }
  30. }
  31. }
  32. int AudioNsImpl::SetNsParams(uint32_t uSampleRate, uint32_t uAudioCapturePeroid, int iMode)
  33. {
  34. int iRet = -1;
  35. m_audio_sample_rate = uSampleRate;
  36. m_audio_capture_peroid = uAudioCapturePeroid;
  37. m_ns_policy = iMode;
  38. if (0 == WebRtcNs_Init(m_NsHandle, uSampleRate)){
  39. iRet = WebRtcNs_set_policy(m_NsHandle, iMode);
  40. if (0 == iRet){
  41. NsLog("WebRtcNs init and set policy success.");
  42. }
  43. else{
  44. NsLog("WebRtcNs set policy failed.");
  45. }
  46. }
  47. else{
  48. NsLog("WebRtcNs Init failed.");
  49. }
  50. NsLog("audio sample rate is %d, audio frame time is %dms, noise suppression policy is %d.",m_audio_sample_rate, m_audio_capture_peroid, m_ns_policy);
  51. return iRet;
  52. }
  53. int AudioNsImpl::NsProcess(char* pDst, uint32_t uDstLen, char* pSrc, uint32_t uSrcLen)
  54. {
  55. int iRet = -1;
  56. switch(m_audio_sample_rate)
  57. {
  58. case 8000:
  59. iRet = NsProcess8kAudioSampleRate(pDst, uDstLen, pSrc, uSrcLen);
  60. break;
  61. case 16000:
  62. iRet = NsProcess16kAudioSampleRate(pDst, uDstLen, pSrc, uSrcLen);
  63. break;
  64. case 32000:
  65. iRet = NsProcess32kAudioSampleRate(pDst, uDstLen, pSrc, uSrcLen);
  66. break;
  67. case 48000:
  68. iRet = NsProcess48kAudioSampleRate(pDst, uDstLen, pSrc, uSrcLen);
  69. break;
  70. default:
  71. NsLog("Rvc_NsProcess not support audio sample rate.");
  72. break;
  73. }
  74. return iRet;
  75. }
  76. //8k采样率,采样间隔20ms,每次传过来音频大小为320字节,10ms缓冲区大小为160字节,WebRtcNs_Process每次处理10ms数据
  77. int AudioNsImpl::NsProcess8kAudioSampleRate(char* pDst, uint32_t uDstLen, char* pSrc, uint32_t uSrcLen)
  78. {
  79. int iRet = -1;
  80. if (NULL == pDst || pSrc == NULL){
  81. return iRet;
  82. }
  83. if ((uDstLen >= 320) && (320 == uSrcLen)){
  84. for (int i = 0; i < m_audio_capture_peroid/RVC_NS_AUDIO_FRAME_TIME; i++){
  85. short shBufferIn[80] = {0};
  86. short shBufferOut[80] = {0};
  87. memcpy(shBufferIn, (char*)pSrc+i*80*sizeof(short), sizeof(short)*80);
  88. if (iRet = WebRtcNs_Process(m_NsHandle ,shBufferIn ,NULL ,shBufferOut , NULL)){
  89. NsLog("Noise_Suppression WebRtcNs_Process 8k input err!");
  90. }
  91. else{
  92. memcpy((char*)pDst+i*80*sizeof(short),shBufferOut,80*sizeof(short));
  93. }
  94. }
  95. }
  96. else if ((160 == uDstLen) && (160 == uSrcLen)){
  97. iRet = WebRtcNs_Process(m_NsHandle, (short*)pSrc, NULL, (short*)pDst, NULL);
  98. }
  99. return iRet;
  100. }
  101. //16k采样率,采样间隔20ms,每次传过来音频大小为640字节,10ms缓冲区大小为320字节,WebRtcNs_Process每次处理10ms数据
  102. int AudioNsImpl::NsProcess16kAudioSampleRate(char* pDst, uint32_t uDstLen, char* pSrc, uint32_t uSrcLen)
  103. {
  104. int iRet = -1;
  105. if (NULL == pDst || pSrc == NULL){
  106. return iRet;
  107. }
  108. if ((640 == uDstLen) && (640 == uSrcLen)){
  109. for (int i = 0; i < m_audio_capture_peroid/RVC_NS_AUDIO_FRAME_TIME; i++){
  110. short shBufferIn[160] = {0};
  111. short shBufferOut[160] = {0};
  112. memcpy(shBufferIn, (char*)pSrc+i*160*sizeof(short), sizeof(short)*160);
  113. if (iRet = WebRtcNs_Process(m_NsHandle ,shBufferIn ,NULL ,shBufferOut , NULL)){
  114. NsLog("Noise_Suppression WebRtcNs_Process 16k input err!");
  115. }
  116. else{
  117. memcpy((char*)pDst+i*160*sizeof(short),shBufferOut,160*sizeof(short));
  118. }
  119. }
  120. }
  121. return iRet;
  122. }
  123. //32k采样率,采样间隔20ms,每次传过来音频大小为1280字节,10ms缓冲区大小为640字节,WebRtcNs_Process每次处理10ms数据
  124. int AudioNsImpl::NsProcess32kAudioSampleRate(char* pDst, uint32_t uDstLen, char* pSrc, uint32_t uSrcLen)
  125. {
  126. int iRet = -1;
  127. if (NULL == pDst || pSrc == NULL){
  128. return iRet;
  129. }
  130. int filter_state1[6] = {0}, filter_state12[6] = {0};
  131. int Synthesis_state1[6] = {0}, Synthesis_state12[6] = {0};
  132. if ((1280 == uDstLen) && (1280 == uSrcLen)){
  133. for (int i = 0; i < m_audio_capture_peroid/RVC_NS_AUDIO_FRAME_TIME; i++){
  134. short shBufferIn[320] = {0};
  135. short shInL[160]={0},shInH[160]={0};
  136. short shOutL[160] ={0},shOutH[160] = {0};
  137. memcpy(shBufferIn, pSrc + i*320*sizeof(short), 320*sizeof(short));
  138. //首先需要使用滤波函数将音频数据分高低频,以高频和低频的方式传入降噪函数内部
  139. WebRtcSpl_AnalysisQMF(shBufferIn,320,shInL,shInH,filter_state1,filter_state12);
  140. //将需要降噪的数据以高频和低频传入对应接口,同时需要注意返回数据也是分高频和低频
  141. if (iRet == WebRtcNs_Process(m_NsHandle, shInL, shInH, shOutL, shOutH)){
  142. NsLog("Noise_Suppression WebRtcNs_Process 32k input err!");
  143. }
  144. else
  145. {
  146. short shBufferOut[320] = {0};
  147. //如果降噪成功,则根据降噪后高频和低频数据传入滤波接口,然后用将返回的数据写入文件
  148. WebRtcSpl_SynthesisQMF(shOutL,shOutH,160,shBufferOut,Synthesis_state1,Synthesis_state12);
  149. memcpy(pDst+i*320*sizeof(short),shBufferOut,320*sizeof(short));
  150. }
  151. }
  152. }
  153. return iRet;
  154. }
  155. //48k采样率,采样间隔20ms,每次传过来音频大小为1920字节,10ms缓冲区大小为960字节,WebRtcNs_Process每次处理10ms数据
  156. int AudioNsImpl::NsProcess48kAudioSampleRate(char* pDst, uint32_t uDstLen, char* pSrc, uint32_t uSrcLen)
  157. {
  158. int iRet = -1;
  159. if (NULL == pDst || pSrc == NULL){
  160. return iRet;
  161. }
  162. int filter_state1[6] = {0}, filter_state12[6] = {0};
  163. int Synthesis_state1[6] = {0}, Synthesis_state12[6] = {0};
  164. if ((uDstLen >= 1920) && (1920 == uSrcLen)){
  165. for (int i = 0; i < m_audio_capture_peroid/RVC_NS_AUDIO_FRAME_TIME; i++){
  166. short shBufferIn[480] = {0};
  167. short shInL[240]={0},shInH[240]={0};
  168. short shOutL[240] ={0},shOutH[240] = {0};
  169. memcpy(shBufferIn, pSrc + i*480*sizeof(short), 480*sizeof(short));
  170. //首先需要使用滤波函数将音频数据分高低频,以高频和低频的方式传入降噪函数内部
  171. WebRtcSpl_AnalysisQMF(shBufferIn,480,shInL,shInH,filter_state1,filter_state12);
  172. //将需要降噪的数据以高频和低频传入对应接口,同时需要注意返回数据也是分高频和低频
  173. if (iRet == WebRtcNs_Process(m_NsHandle, shInL, shInH, shOutL, shOutH)){
  174. NsLog("Noise_Suppression WebRtcNs_Process 48k input err!");
  175. }
  176. else
  177. {
  178. short shBufferOut[480] = {0};
  179. //如果降噪成功,则根据降噪后高频和低频数据传入滤波接口,然后用将返回的数据写入文件
  180. WebRtcSpl_SynthesisQMF(shOutL,shOutH,240,shBufferOut,Synthesis_state1,Synthesis_state12);
  181. memcpy(pDst+i*480*sizeof(short),shBufferOut,480*sizeof(short));
  182. }
  183. }
  184. }
  185. return iRet;
  186. }
  187. void AudioNsImpl::NsLog(const char* fmt, ...)
  188. {
  189. if (m_nscallback.debug){
  190. va_list arg;
  191. va_start(arg, fmt);
  192. (*m_nscallback.debug)(m_nscallback.user_data, fmt, arg);
  193. va_end(arg);
  194. }
  195. }
  196. void AudioNsImpl::ReleaseObj()
  197. {
  198. NsLog("AudioNsImpl ReleaseObj.");
  199. delete this;
  200. }