Adding Opus to RTPencode.

webrtc/modules/audio_coding/neteq/test/RTPencode.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 // TODO(hlundin): Reformat file to meet style guide.
 
 /* header includes */
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #ifdef WIN32
 #include <winsock2.h>
 #endif
 #ifdef WEBRTC_LINUX
 #include <netinet/in.h>
 #endif
 
 #include <assert.h>
 
 #include <algorithm>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/typedefs.h"
+
 // needed for NetEqDecoder
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 
 /************************/
 /* Define payload types */
 /************************/
 
 #include "PayloadTypes.h"
 
+namespace {
+const size_t kRtpDataSize = 8000;
+}
+
 /*********************/
 /* Misc. definitions */
 /*********************/
 
 #define STOPSENDTIME 3000
 #define RESTARTSENDTIME 0  // 162500
 #define FIRSTLINELEN 40
 #define CHECK_NOT_NULL(a)                                                \
   if ((a) == 0) {                                                        \
     printf("\n %s \n line: %d \nerror at %s\n", __FILE__, __LINE__, #a); \
@@ skipping 137 matching lines @@
 #include "GSMFRInterface.h"
 #include "GSMFRCreation.h"
 #endif
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
 #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 #endif
 #if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
 #include "SpeexInterface.h"
 #endif
+#ifdef CODEC_OPUS
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#endif
 
 /***********************************/
 /* Global codec instance variables */
 /***********************************/
 
 WebRtcVadInst* VAD_inst[2];
 
 #ifdef CODEC_G722
 G722EncInst* g722EncState[2];
 #endif
@@ skipping 51 matching lines @@
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
 CNG_enc_inst* CNGenc_inst[2];
 #endif
 #ifdef CODEC_SPEEX_8
 SPEEX_encinst_t* SPEEX8enc_inst[2];
 #endif
 #ifdef CODEC_SPEEX_16
 SPEEX_encinst_t* SPEEX16enc_inst[2];
 #endif
+#ifdef CODEC_OPUS
+OpusEncInst* opus_inst[2];
+#endif
 
 int main(int argc, char* argv[]) {
   size_t packet_size;
   int fs;
   webrtc::NetEqDecoder usedCodec;
   int payloadType;
   int bitrate = 0;
   int useVAD, vad;
   int useRed = 0;
   size_t len, enc_len;
   int16_t org_data[4000];
-  unsigned char rtp_data[8000];
+  unsigned char rtp_data[kRtpDataSize];
   int16_t seqNo = 0xFFF;
   uint32_t ssrc = 1235412312;
   uint32_t timestamp = 0xAC1245;
   uint16_t length, plen;
   uint32_t offset;
   double sendtime = 0;
   int red_PT[2] = {0};
   uint32_t red_TS[2] = {0};
   uint16_t red_len[2] = {0};
   size_t RTPheaderLen = 12;
-  uint8_t red_data[8000];
+  uint8_t red_data[kRtpDataSize];
 #ifdef INSERT_OLD_PACKETS
   uint16_t old_length, old_plen;
   size_t old_enc_len;
   int first_old_packet = 1;
-  unsigned char old_rtp_data[8000];
+  unsigned char old_rtp_data[kRtpDataSize];
   size_t packet_age = 0;
 #endif
 #ifdef INSERT_DTMF_PACKETS
   int NTone = 1;
   int DTMFfirst = 1;
   uint32_t DTMFtimestamp;
   bool dtmfSent = false;
 #endif
   bool usingStereo = false;
   size_t stereoMode = 0;
@@ skipping 117 matching lines @@
 #endif
 #ifdef CODEC_RED
 #ifdef CODEC_G711
     printf("             : red_pcm      Redundancy RTP packet with 2*G711A "
            "frames\n");
 #endif
 #ifdef CODEC_ISAC
     printf("             : red_isac     Redundancy RTP packet with 2*iSAC "
            "frames\n");
 #endif
+#endif  // CODEC_RED
+#ifdef CODEC_OPUS
+    printf("             : opus         Opus codec with FEC (48kHz, 32kbps, FEC"
+        " on and tuned for 5%% packet losses)\n");
 #endif
     printf("\n");
 
 #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
      defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
     printf("useVAD       : 0 Voice Activity Detection is switched off\n");
     printf("             : 1 Voice Activity Detection is switched on\n\n");
 #else
     printf("useVAD       : 0 Voice Activity Detection switched off (on not "
            "supported)\n\n");
@@ skipping 431 matching lines @@
   } else if (!strcmp(name, "red_pcm")) {
     *codec = webrtc::NetEqDecoder::kDecoderPCMa;
     *PT = NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
     *fs = 8000;
     *useRed = 1;
   } else if (!strcmp(name, "red_isac")) {
     *codec = webrtc::NetEqDecoder::kDecoderISAC;
     *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
     *fs = 16000;
     *useRed = 1;
+  } else if (!strcmp(name, "opus")) {
+    *codec = webrtc::NetEqDecoder::kDecoderOpus;
+    *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */
+    *fs = 48000;
   } else {
     printf("Error: Not a supported codec (%s)\n", name);
     exit(0);
   }
 }
 
 int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
                           size_t enc_frameSize,
                           int bitrate,
                           int sampfreq,
@@ skipping 511 matching lines @@
                    "enc_frameSize\n");
             exit(0);
           }
           WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
         } else {
           printf("\nError - GSM FR is only developed for 8kHz \n");
           exit(0);
         }
         break;
 #endif
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0);
+        if (ok != 0) {
+          printf("Error: Couldn't allocate memory for Opus encoding "
+                 "instance\n");
+          exit(0);
+        }
+        WebRtcOpus_EnableFec(opus_inst[k]);
+        WebRtcOpus_SetPacketLossRate(opus_inst[k], 5);
+        break;
+#endif
       default:
         printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
         exit(0);
         break;
     }
-
     if (ok != 0) {
       return (ok);
     }
   }  // end for
 
   return (0);
 }
 
 int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
   for (size_t k = 0; k < numChannels; k++) {
@@ skipping 106 matching lines @@
 #ifdef CODEC_ISAC_SWB
       case webrtc::NetEqDecoder::kDecoderISACswb:
         WebRtcIsac_Free(ISACSWB_inst[k]);
         break;
 #endif
 #ifdef CODEC_GSMFR
       case webrtc::NetEqDecoder::kDecoderGSMFR:
         WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
         break;
 #endif
+#ifdef CODEC_OPUS
+      case webrtc::NetEqDecoder::kDecoderOpus:
+        WebRtcOpus_EncoderFree(opus_inst[k]);
+        break;
+#endif
       default:
         printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
         exit(0);
         break;
     }
   }
 
   return (0);
 }
 
@@ skipping 124 matching lines @@
       int noOfCalls = 0;
       int res = 0;
       while (res <= 0) {
         res = WebRtcIsac_Encode(ISACSWB_inst[k], &indata[noOfCalls * 320],
                                 encoded);
         noOfCalls++;
       }
       cdlen = static_cast<size_t>(res);
     }
 #endif
+#ifdef CODEC_OPUS
+    cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12,
+                              encoded);
+    RTC_CHECK_GT(cdlen, 0u);
+#endif
     indata += frameLen;
     encoded += cdlen;
     totalLen += cdlen;
 
   }  // end for
 
   first_cng = 1;
   return (totalLen);
 }
 
@@ skipping 137 matching lines @@
     memmove(ptrL + stride, ptrL, ptrR - ptrL);
 
     // copy from temp to left pointer
     memcpy(ptrL, temp, stride);
 
     // advance pointers
     ptrL += stride * 2;
     ptrR += stride;
   }
 }