76c53d36bc
References channels and underlying objects directly instead of using interfaces referenced with channel id. Channel creation is still done as before for now. BUG=1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46849004 Cr-Commit-Position: refs/heads/master@{#8958}
142 lines
4.0 KiB
C++
142 lines
4.0 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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class LoadObserver {
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public:
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enum Load { kOveruse, kUnderuse };
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// Triggered when overuse is detected or when we believe the system can take
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// more load.
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virtual void OnLoadUpdate(Load load) = 0;
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protected:
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virtual ~LoadObserver() {}
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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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: send_transport(send_transport),
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voice_engine(NULL),
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overuse_callback(NULL) {}
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static const int kDefaultStartBitrateBps;
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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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LoadObserver* overuse_callback;
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// Bitrate config used until valid bitrate estimates are calculated. Also
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// used to cap total bitrate used.
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struct BitrateConfig {
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BitrateConfig()
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: min_bitrate_bps(0),
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start_bitrate_bps(kDefaultStartBitrateBps),
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max_bitrate_bps(-1) {}
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int min_bitrate_bps;
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int start_bitrate_bps;
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int max_bitrate_bps;
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} bitrate_config;
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};
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struct Stats {
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Stats()
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: send_bandwidth_bps(0),
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recv_bandwidth_bps(0),
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pacer_delay_ms(0),
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rtt_ms(-1) {}
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int send_bandwidth_bps;
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int recv_bandwidth_bps;
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int64_t pacer_delay_ms;
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int64_t rtt_ms;
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};
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static Call* Create(const Call::Config& 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 call statistics, such as estimated send and receive bandwidth,
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// pacing delay, etc.
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virtual Stats GetStats() const = 0;
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// TODO(pbos): Like BitrateConfig above this is currently per-stream instead
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// of maximum for entire Call. This should be fixed along with the above.
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// Specifying a start bitrate (>0) will currently reset the current bitrate
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// estimate. This is due to how the 'x-google-start-bitrate' flag is currently
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// implemented.
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virtual void SetBitrateConfig(
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const Config::BitrateConfig& bitrate_config) = 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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