caba2d2a37
Allows signalling why packet delivery failed. Especially enables signaling that delivery fails because the incoming packet had an unknown SSRC. This allows an application to react and create receivers for the new streams. R=mflodman@webrtc.org BUG=3228 Review URL: https://webrtc-codereview.appspot.com/12289005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
115 lines
3.4 KiB
C++
115 lines
3.4 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_CALL_H_
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#define WEBRTC_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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enum DeliveryStatus {
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DELIVERY_OK,
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DELIVERY_UNKNOWN_SSRC,
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DELIVERY_PACKET_ERROR,
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};
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virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
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size_t length) = 0;
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protected:
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virtual ~PacketReceiver() {}
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};
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// Callback interface for reporting when a system overuse is detected.
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// The detection is based on the jitter of incoming captured frames.
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class OveruseCallback {
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public:
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// Called as soon as an overuse is detected.
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virtual void OnOveruse() = 0;
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// Called periodically when the system is not overused any longer.
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virtual void OnNormalUse() = 0;
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protected:
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virtual ~OveruseCallback() {}
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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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: webrtc_config(NULL),
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send_transport(send_transport),
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voice_engine(NULL),
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overuse_callback(NULL) {}
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webrtc::Config* webrtc_config;
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newapi::Transport* send_transport;
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// VoiceEngine used for audio/video synchronization for this Call.
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VoiceEngine* voice_engine;
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// Callback for overuse and normal usage based on the jitter of incoming
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// captured frames. 'NULL' disables the callback.
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OveruseCallback* overuse_callback;
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};
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static Call* Create(const Call::Config& config);
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static Call* Create(const Call::Config& config,
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const webrtc::Config& webrtc_config);
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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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virtual void DestroyVideoSendStream(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 DestroyVideoReceiveStream(
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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_CALL_H_
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