Rename VideoCall to Call.
Call should encompass more than video, there's no point in calling it
VideoCall.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/new_include/call.h b/video_engine/new_include/call.h
new file mode 100644
index 0000000..6650216
--- /dev/null
+++ b/video_engine/new_include/call.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (c) 2013 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.
+ */
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/video_engine/new_include/video_receive_stream.h"
+#include "webrtc/video_engine/new_include/video_send_stream.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+const char* Version();
+
+class PacketReceiver {
+ public:
+ virtual bool DeliverPacket(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+ virtual ~PacketReceiver() {}
+};
+
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+class Call {
+ public:
+ struct Config {
+ explicit Config(newapi::Transport* send_transport)
+ : send_transport(send_transport),
+ overuse_detection(false),
+ voice_engine(NULL),
+ trace_callback(NULL),
+ trace_filter(kTraceNone) {}
+
+ newapi::Transport* send_transport;
+ bool overuse_detection;
+
+ // VoiceEngine used for audio/video synchronization for this Call.
+ VoiceEngine* voice_engine;
+
+ TraceCallback* trace_callback;
+ uint32_t trace_filter;
+ };
+
+ static Call* Create(const Call::Config& config);
+
+ virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
+
+ virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
+
+ virtual VideoSendStream* CreateSendStream(
+ const VideoSendStream::Config& config) = 0;
+
+ // Returns the internal state of the send stream, for resume sending with a
+ // new stream with different settings.
+ // Note: Only the last returned send-stream state is valid.
+ virtual SendStreamState* DestroySendStream(VideoSendStream* send_stream) = 0;
+
+ virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
+
+ virtual VideoReceiveStream* CreateReceiveStream(
+ const VideoReceiveStream::Config& config) = 0;
+ virtual void DestroyReceiveStream(VideoReceiveStream* receive_stream) = 0;
+
+ // All received RTP and RTCP packets for the call should be inserted to this
+ // PacketReceiver. The PacketReceiver pointer is valid as long as the
+ // Call instance exists.
+ virtual PacketReceiver* Receiver() = 0;
+
+ // Returns the estimated total send bandwidth. Note: this can differ from the
+ // actual encoded bitrate.
+ virtual uint32_t SendBitrateEstimate() = 0;
+
+ // Returns the total estimated receive bandwidth for the call. Note: this can
+ // differ from the actual receive bitrate.
+ virtual uint32_t ReceiveBitrateEstimate() = 0;
+
+ virtual ~Call() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_