audioEnabled = false

ljc2266 · Nov 6, 2016 · 0027241 · 0027241
1 parent 99e2a27
commit 0027241
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Showing 6 changed files with 77 additions and 18 deletions.
diff --git a/WebRtc.NET/src/conductor.cc b/WebRtc.NET/src/conductor.cc
@@ -21,6 +21,7 @@
 
 namespace Native
 {
+	const char kAudioLabel[] = "audio_label";
 	const char kVideoLabel[] = "video_label";
 	const char kStreamLabel[] = "stream_label";
 	const char kSoftware[] = "libjingle TurnServer";
@@ -62,8 +63,11 @@ namespace Native
 		onFailure = nullptr;
 		onIceCandidate = nullptr;
 		capturer = nullptr;
+
 		caputureFps = 5;
 		barcodeEnabled = false;
+		audioEnabled = false;
+
 		turnServer = nullptr;
 		data_channel = nullptr;
 		onDataMessage = nullptr;
@@ -98,7 +102,7 @@ namespace Native
 	{
 		if (peer_connection_)
 		{
-			local_renderer_.reset();
+			local_video.reset();
 
 			for (auto it = active_streams_.begin(); it != active_streams_.end(); ++it)
 			{
@@ -309,25 +313,26 @@ namespace Native
 		if (active_streams_.find(kStreamLabel) != active_streams_.end())
 			return;  // Already added.
 
-		//webrtc::AudioTrackInterface * audio_track = peer_connection_factory_->CreateAudioTrack(
-		//		kAudioLabel, peer_connection_factory_->CreateAudioSource(NULL));
-
 		if (!capturer)
 		{
 			capturer = new Native::YuvFramesCapturer2(*this);
 		}
 
 		auto v = pc_factory_->CreateVideoSource(capturer, NULL);
-
 		auto video_track = pc_factory_->CreateVideoTrack(kVideoLabel, v);
 		if (onRenderLocal)
 		{
-			local_renderer_.reset(new VideoRenderer(*this, false, video_track));
+			local_video.reset(new VideoRenderer(*this, false, video_track));
 	    }
 
 		auto stream = pc_factory_->CreateLocalMediaStream(kStreamLabel);
 		{
-			//stream->AddTrack(audio_track);
+			if (audioEnabled)
+			{
+				auto a = pc_factory_->CreateAudioSource(NULL);
+				auto audio_track = pc_factory_->CreateAudioTrack(kAudioLabel, a);
+				stream->AddTrack(audio_track);
+			}
 			stream->AddTrack(video_track);
 
 			if (!peer_connection_->AddStream(stream))
@@ -347,19 +352,29 @@ namespace Native
 		if (onRenderRemote)
 		{
 			webrtc::VideoTrackVector tracks = stream->GetVideoTracks();
-			// Only render the first track.
 			if (!tracks.empty())
 			{
 				webrtc::VideoTrackInterface* track = tracks[0];
-				remote_renderer_.reset(new VideoRenderer(*this, true, track));
+				remote_video.reset(new Native::VideoRenderer(*this, true, track));
+			}
+		}
+
+		if (audioEnabled)
+		{
+			webrtc::AudioTrackVector atracks = stream->GetAudioTracks();
+			if (!atracks.empty())
+			{
+				webrtc::AudioTrackInterface* track = atracks[0];
+				remote_audio.reset(new Native::AudioRenderer(*this, true, track));
 			}
 		}
 	}
 
 	void Conductor::OnRemoveStream(rtc::scoped_refptr<webrtc::MediaStreamInterface> stream)
 	{
 		LOG(INFO) << __FUNCTION__ << " " << stream->label();
-		remote_renderer_.reset();
+		remote_video.reset();
+		remote_audio.reset();
 	}
 
 	void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate)

diff --git a/WebRtc.NET/src/conductor.h b/WebRtc.NET/src/conductor.h
@@ -166,15 +166,17 @@ namespace Native
 
 		std::vector<webrtc::PeerConnectionInterface::IceServer> serverConfigs;
 
-		std::unique_ptr<VideoRenderer> local_renderer_;
-		std::unique_ptr<VideoRenderer> remote_renderer_;
+		std::unique_ptr<VideoRenderer> local_video;
+		std::unique_ptr<VideoRenderer> remote_video;
+		std::unique_ptr<AudioRenderer> remote_audio;
 
 		cricket::TurnServer * turnServer;
 		cricket::StunServer * stunServer;
 
 	public:
 		int caputureFps;
 		bool barcodeEnabled;
+		bool audioEnabled;
 	};
 }
 #endif  // WEBRTC_NET_CONDUCTOR_H_
diff --git a/WebRtc.NET/src/defaults.cc b/WebRtc.NET/src/defaults.cc
@@ -214,4 +214,20 @@ namespace Native
 			con->onRenderLocal((uint8_t*)b->DataY(), b->width(), b->height());
 		}
 	}
+
+	// AudioTrackSinkInterface implementation
+	void AudioRenderer::OnData(const void* audio_data,
+							   int bits_per_sample,
+							   int sample_rate,
+							   size_t number_of_channels,
+							   size_t number_of_frames)
+	{
+		std::stringstream s;
+		s << "AudioRenderer::OnData, bps: " << bits_per_sample 
+		<< ", rate: " << sample_rate 
+		<< ", channels: " << number_of_channels
+		<< ", frames: " << number_of_frames;
+
+		::OutputDebugStringA(s.str().c_str());
+	}
 }
diff --git a/WebRtc.NET/src/defaults.h b/WebRtc.NET/src/defaults.h
@@ -74,8 +74,33 @@ namespace Native
 
 		bool remote;
 		Conductor * con;
-		CRITICAL_SECTION buffer_lock_;
-		rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_;
+		rtc::scoped_refptr<webrtc::VideoTrackInterface> rendered_track_;		
 	};
+
+	class AudioRenderer : public webrtc::AudioTrackSinkInterface
+	{
+	public:
+		AudioRenderer(Conductor & c, bool remote, webrtc::AudioTrackInterface * track_to_render) :
+			audio_track_(track_to_render), con(&c), remote(remote)
+		{
+			audio_track_->AddSink(this);
+		}
+		virtual ~AudioRenderer()
+		{
+			audio_track_->RemoveSink(this);
+		}
+
+		virtual void OnData(const void* audio_data,
+							int bits_per_sample,
+							int sample_rate,
+							size_t number_of_channels,
+							size_t number_of_frames) override;
+
+	protected:
+
+		bool remote;
+		Conductor * con;
+		rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track_;
+	};	
 }
 #endif  // WEBRTC_NET_DEFAULTS_H_
diff --git a/web/index.html b/web/index.html
@@ -81,7 +81,7 @@
             <div style="color: white" id="statsdiv"></div>
         </div>
         </br>
-        <video id="vidLocal" autoplay controls style="display: none"></video>
+        <video id="vidLocal" autoplay controls muted style="display: none"></video>
     </div>
 </body >
 </html >
diff --git a/web/rtc_main.js b/web/rtc_main.js
@@ -37,7 +37,8 @@ var offerOptions = {
 };
 
 var vgaConstraints = {
-    video: true
+    video: true,
+    //audio: true
 };
 
 var dataChannelOptions = {
@@ -111,15 +112,15 @@ function startStream() {
                                 var s = o[key];
                                 return (
                                         (s.type == "inboundrtp" && !s.isRemote) ||
-                                        (s.type == "ssrc" && s.id.indexOf("recv") >= 0)
+                                        (s.type == "ssrc" && s.mediaType == "video" && s.id.indexOf("recv") >= 0)
                                        );
                             })];
 
                             var snd = o[Object.keys(o).find(function (key) {
                                 var s = o[key];
                                 return (
                                         (s.type == "outboundrtp" && !s.isRemote) ||
-                                        (s.type == "ssrc" && s.id.indexOf("send") >= 0)
+                                        (s.type == "ssrc" && s.mediaType == "video" && s.id.indexOf("send") >= 0)
                                        );
                             })];