Support OPUS

README.md

@@ -10,8 +10,9 @@ libpeer is a WebRTC implementation written in C, developed with BSD socket. The
   - H264
   - G.711 PCM (A-law)
   - G.711 PCM (µ-law)
+  - OPUS
 - DataChannel
-- STUN
+- STUN/TURN
 - Signaling
 
 ### Dependencies

src/config.h

@@ -9,12 +9,14 @@
 #define VIDEO_RB_DATA_LENGTH (CONFIG_MTU * 64)
 #define AUDIO_RB_DATA_LENGTH (CONFIG_MTU * 64)
 #define DATA_RB_DATA_LENGTH (SCTP_MTU * 128)
+#define AUDIO_LATENCY 50 // ms
 #else
 #define HAVE_USRSCTP
 #define RSA_KEY_LENGTH 2048
 #define VIDEO_RB_DATA_LENGTH (CONFIG_MTU * 256)
 #define AUDIO_RB_DATA_LENGTH (CONFIG_MTU * 256)
 #define DATA_RB_DATA_LENGTH (SCTP_MTU * 128)
+#define AUDIO_LATENCY 20 // ms
 #endif
 
 // siganling

src/peer_connection.c

@@ -308,6 +308,13 @@ static void peer_connection_state_new(PeerConnection *pc) {
       sdp_append(&pc->local_sdp, "a=setup:actpass");
       strcat(pc->local_sdp.content, description);
       break;
+
+    case CODEC_OPUS:
+      sdp_append_opus(&pc->local_sdp);
+      sdp_append(&pc->local_sdp, "a=fingerprint:sha-256 %s", pc->dtls_srtp.local_fingerprint);
+      sdp_append(&pc->local_sdp, "a=setup:actpass");
+      strcat(pc->local_sdp.content, description);
+
     default:
       break;
   }

src/rtp.c

@@ -77,7 +77,7 @@ static int rtp_encoder_encode_h264_fu_a(RtpEncoder *rtp_encoder, uint8_t *buf, s
   rtp_packet->header.seq_number = htons(rtp_encoder->seq_number++);
   rtp_packet->header.timestamp = htonl(rtp_encoder->timestamp);
   rtp_packet->header.ssrc = htonl(rtp_encoder->ssrc);
-  rtp_encoder->timestamp += 90000/25; // 25 FPS.
+  rtp_encoder->timestamp += rtp_encoder->timestamp_increment;
 
   uint8_t type = buf[0] & 0x1f;
   uint8_t nri = (buf[0] & 0x60) >> 5;
@@ -173,7 +173,7 @@ static int rtp_encoder_encode_generic(RtpEncoder *rtp_encoder, uint8_t *buf, siz
   rtp_header->markerbit = 0;
   rtp_header->type = rtp_encoder->type;
   rtp_header->seq_number = htons(rtp_encoder->seq_number++);
-  rtp_encoder->timestamp += size; // 8000 HZ. 
+  rtp_encoder->timestamp += rtp_encoder->timestamp_increment;
   rtp_header->timestamp = htonl(rtp_encoder->timestamp);
   rtp_header->ssrc = htonl(rtp_encoder->ssrc);
   memcpy(rtp_encoder->buf + sizeof(RtpHeader), buf, size);
@@ -195,16 +195,24 @@ void rtp_encoder_init(RtpEncoder *rtp_encoder, MediaCodec codec, RtpOnPacket on_
     case CODEC_H264:
       rtp_encoder->type = PT_H264;
       rtp_encoder->ssrc = SSRC_H264;
+      rtp_encoder->timestamp_increment = 90000/30; // 25 FPS.
       rtp_encoder->encode_func = rtp_encoder_encode_h264;
       break;
     case CODEC_PCMA:
       rtp_encoder->type = PT_PCMA;
       rtp_encoder->ssrc = SSRC_PCMA;
+      rtp_encoder->timestamp_increment = AUDIO_LATENCY*8000/1000;
       rtp_encoder->encode_func = rtp_encoder_encode_generic;
       break;
     case CODEC_PCMU:
       rtp_encoder->type = PT_PCMU;
       rtp_encoder->ssrc = SSRC_PCMU;
+      rtp_encoder->timestamp_increment = AUDIO_LATENCY*8000/1000;
+      rtp_encoder->encode_func = rtp_encoder_encode_generic;
+    case CODEC_OPUS:
+      rtp_encoder->type = PT_OPUS;
+      rtp_encoder->ssrc = SSRC_OPUS;
+      rtp_encoder->timestamp_increment = AUDIO_LATENCY*48000/1000;
       rtp_encoder->encode_func = rtp_encoder_encode_generic;
     default:
       break;
@@ -235,9 +243,8 @@ void rtp_decoder_init(RtpDecoder *rtp_decoder, MediaCodec codec, RtpOnPacket on_
       rtp_decoder->decode_func = NULL;
       break;
     case CODEC_PCMA:
-      rtp_decoder->decode_func = rtp_decode_generic;
-      break;
     case CODEC_PCMU:
+    case CODEC_OPUS:
       rtp_decoder->decode_func = rtp_decode_generic;
     default:
       break;

src/rtp.h

@@ -91,6 +91,7 @@ struct RtpEncoder {
   uint16_t seq_number;
   uint32_t ssrc;
   uint32_t timestamp;
+  uint32_t timestamp_increment;
   uint8_t buf[CONFIG_MTU + 1];
 };
 

src/sdp.c

@@ -65,6 +65,17 @@ void sdp_append_pcmu(Sdp *sdp) {
   sdp_append(sdp, "a=rtcp-mux");
 }
 
+void sdp_append_opus(Sdp *sdp) {
+
+  sdp_append(sdp, "m=audio 9 UDP/TLS/RTP/SAVP 111");
+  sdp_append(sdp, "a=rtpmap:111 opus/48000/2");
+  sdp_append(sdp, "a=ssrc:6 cname:webrtc-opus");
+  sdp_append(sdp, "a=sendrecv");
+  sdp_append(sdp, "a=mid:audio");
+  sdp_append(sdp, "c=IN IP4 0.0.0.0");
+  sdp_append(sdp, "a=rtcp-mux");
+}
+
 void sdp_append_datachannel(Sdp *sdp) {
 
   sdp_append(sdp, "m=application 50712 UDP/DTLS/SCTP webrtc-datachannel");

src/sdp.h

@@ -18,6 +18,8 @@ void sdp_append_pcma(Sdp *sdp);
 
 void sdp_append_pcmu(Sdp *sdp);
 
+void sdp_append_opus(Sdp *sdp);
+
 void sdp_append_datachannel(Sdp *sdp);
 
 void sdp_create(Sdp *sdp, int b_video, int b_audio, int b_datachannel);