bbe0a8517d
Used for config parameters in common between multiple codecs as well as the encoder-specific pointer. In particular this contains content mode (realtime video vs. screenshare). BUG=1788 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
3.6 KiB
C++
124 lines
3.6 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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enum NetworkState {
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kNetworkUp,
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kNetworkDown,
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};
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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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start_bitrate_bps(-1) {}
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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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// Start bitrate used before a valid bitrate estimate is calculated. '-1'
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// lets the call decide start bitrate.
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// Note: This currently only affects video.
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int start_bitrate_bps;
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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* CreateVideoSendStream(
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const VideoSendStream::Config& config,
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const VideoEncoderConfig& encoder_config) = 0;
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virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 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 void SignalNetworkState(NetworkState state) = 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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