5a63655ab0
Renaming the methods to include Video. Long-term there will hopefully be AudioSendStream/AudioReceiveStreams as well. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
2.9 KiB
C++
95 lines
2.9 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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#include <string>
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#include <vector>
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#include "webrtc/common_types.h"
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#include "webrtc/video_receive_stream.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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class VoiceEngine;
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const char* Version();
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class PacketReceiver {
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public:
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virtual bool DeliverPacket(const uint8_t* packet, size_t length) = 0;
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protected:
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virtual ~PacketReceiver() {}
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};
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// A Call instance can contain several send and/or receive streams. All streams
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// are assumed to have the same remote endpoint and will share bitrate estimates
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// etc.
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class Call {
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public:
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struct Config {
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explicit Config(newapi::Transport* send_transport)
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: send_transport(send_transport),
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overuse_detection(false),
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voice_engine(NULL),
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trace_callback(NULL),
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trace_filter(kTraceDefault) {}
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newapi::Transport* send_transport;
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bool overuse_detection;
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// VoiceEngine used for audio/video synchronization for this Call.
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VoiceEngine* voice_engine;
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TraceCallback* trace_callback;
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uint32_t trace_filter;
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};
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static Call* Create(const Call::Config& config);
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virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
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virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
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virtual VideoSendStream* CreateVideoSendStream(
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const VideoSendStream::Config& config) = 0;
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// Returns the internal state of the send stream, for resume sending with a
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// new stream with different settings.
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// Note: Only the last returned send-stream state is valid.
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virtual void DestroySendStream(VideoSendStream* send_stream) = 0;
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virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
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virtual VideoReceiveStream* CreateVideoReceiveStream(
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const VideoReceiveStream::Config& config) = 0;
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virtual void DestroyReceiveStream(VideoReceiveStream* receive_stream) = 0;
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// All received RTP and RTCP packets for the call should be inserted to this
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// PacketReceiver. The PacketReceiver pointer is valid as long as the
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// Call instance exists.
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virtual PacketReceiver* Receiver() = 0;
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// Returns the estimated total send bandwidth. Note: this can differ from the
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// actual encoded bitrate.
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virtual uint32_t SendBitrateEstimate() = 0;
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// Returns the total estimated receive bandwidth for the call. Note: this can
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// differ from the actual receive bitrate.
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virtual uint32_t ReceiveBitrateEstimate() = 0;
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virtual ~Call() {}
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
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