Adding Opus to RTPencode.

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

Index: webrtc/modules/audio_coding/neteq/test/RTPencode.cc
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 5410925e8100071057f3be86caf87f5a86f17fc6..b199c37c8558923684b9bb90f4e120eb85ab135f 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -36,6 +36,10 @@
 
 #include "PayloadTypes.h"
 
+namespace {
+const size_t kRtpDataSize = 8000;
+}
+
 /*********************/
 /* Misc. definitions */
 /*********************/
@@ -193,6 +197,9 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
 #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 */
@@ -264,6 +271,9 @@ SPEEX_encinst_t* SPEEX8enc_inst[2];
 #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;
@@ -275,7 +285,7 @@ int main(int argc, char* argv[]) {
   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;
@@ -286,12 +296,12 @@ int main(int argc, char* argv[]) {
   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
@@ -429,6 +439,10 @@ int main(int argc, char* argv[]) {
     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");
 
@@ -880,6 +894,10 @@ void NetEQTest_GetCodec_and_PT(char* name,
     *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);
@@ -1411,12 +1429,23 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
         }
         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);
     }
@@ -1543,6 +1572,11 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
         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);
@@ -1687,6 +1721,14 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
       cdlen = static_cast<size_t>(res);
     }
 #endif
+#ifdef CODEC_OPUS
+    cdlen = WebRtcOpus_Encode(opus_inst[k],
+                              indata,
+                              frameLen,
+                              kRtpDataSize - 12,
+                              encoded);
+    assert(cdlen > 0);

[hlundin-webrtc, 2015/12/11 08:30:29]

Nit: CHECK_GT(cdlen, 0);
I think CHECK is better than DCHECK.
Also include checks.h.

[minyue-webrtc, 2015/12/11 09:03:55]

Done.

+#endif
     indata += frameLen;
     encoded += cdlen;
     totalLen += cdlen;