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/*
* 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_CALL_H_
#define WEBRTC_CALL_H_
#include <string>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
class VoiceEngine;
const char* Version();
class PacketReceiver {
public:
enum DeliveryStatus {
DELIVERY_OK,
DELIVERY_UNKNOWN_SSRC,
DELIVERY_PACKET_ERROR,
};
virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
size_t length) = 0;
protected:
virtual ~PacketReceiver() {}
};
// Callback interface for reporting when a system overuse is detected.
// The detection is based on the jitter of incoming captured frames.
class OveruseCallback {
public:
// Called as soon as an overuse is detected.
virtual void OnOveruse() = 0;
// Called periodically when the system is not overused any longer.
virtual void OnNormalUse() = 0;
protected:
virtual ~OveruseCallback() {}
};
// 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)
: webrtc_config(NULL),
send_transport(send_transport),
voice_engine(NULL),
overuse_callback(NULL),
start_bitrate_bps(-1) {}
webrtc::Config* webrtc_config;
newapi::Transport* send_transport;
// VoiceEngine used for audio/video synchronization for this Call.
VoiceEngine* voice_engine;
// Callback for overuse and normal usage based on the jitter of incoming
// captured frames. 'NULL' disables the callback.
OveruseCallback* overuse_callback;
// Start bitrate used before a valid bitrate estimate is calculated. '-1'
// lets the call decide start bitrate.
// Note: This currently only affects video.
int start_bitrate_bps;
};
static Call* Create(const Call::Config& config);
static Call* Create(const Call::Config& config,
const webrtc::Config& webrtc_config);
virtual VideoSendStream* CreateVideoSendStream(
const VideoSendStream::Config& config,
const std::vector<VideoStream>& video_streams,
const void* encoder_settings) = 0;
virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
virtual VideoReceiveStream* CreateVideoReceiveStream(
const VideoReceiveStream::Config& config) = 0;
virtual void DestroyVideoReceiveStream(
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_CALL_H_
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