blob: 033e1a20db95f1fe8c11efe94f78cf412362ab16 [file] [log] [blame]
* 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/audio_receive_stream.h"
#include "webrtc/audio_send_stream.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
class AudioDeviceModule;
class AudioProcessing;
class VoiceEngine;
class VoiceEngineObserver;
const char* Version();
enum class MediaType {
class PacketReceiver {
enum DeliveryStatus {
virtual DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) = 0;
virtual ~PacketReceiver() {}
// Callback interface for reporting when a system overuse is detected.
class LoadObserver {
enum Load { kOveruse, kUnderuse };
// Triggered when overuse is detected or when we believe the system can take
// more load.
virtual void OnLoadUpdate(Load load) = 0;
virtual ~LoadObserver() {}
// 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 {
struct Config {
static const int kDefaultStartBitrateBps;
// VoiceEngine used for audio/video synchronization for this Call.
VoiceEngine* voice_engine = nullptr;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used.
struct BitrateConfig {
int min_bitrate_bps = 0;
int start_bitrate_bps = kDefaultStartBitrateBps;
int max_bitrate_bps = -1;
} bitrate_config;
struct AudioConfig {
AudioDeviceModule* audio_device_module = nullptr;
AudioProcessing* audio_processing = nullptr;
VoiceEngineObserver* voice_engine_observer = nullptr;
} audio_config;
struct Stats {
int send_bandwidth_bps = 0;
int recv_bandwidth_bps = 0;
int64_t pacer_delay_ms = 0;
int64_t rtt_ms = -1;
static Call* Create(const Call::Config& config);
virtual AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) = 0;
virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
virtual AudioReceiveStream* CreateAudioReceiveStream(
const AudioReceiveStream::Config& config) = 0;
virtual void DestroyAudioReceiveStream(
AudioReceiveStream* receive_stream) = 0;
virtual VideoSendStream* CreateVideoSendStream(
const VideoSendStream::Config& config,
const VideoEncoderConfig& encoder_config) = 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 call statistics, such as estimated send and receive bandwidth,
// pacing delay, etc.
virtual Stats GetStats() const = 0;
// TODO(pbos): Like BitrateConfig above this is currently per-stream instead
// of maximum for entire Call. This should be fixed along with the above.
// Specifying a start bitrate (>0) will currently reset the current bitrate
// estimate. This is due to how the 'x-google-start-bitrate' flag is currently
// implemented.
virtual void SetBitrateConfig(
const Config::BitrateConfig& bitrate_config) = 0;
virtual void SignalNetworkState(NetworkState state) = 0;
virtual ~Call() {}
} // namespace webrtc
#endif // WEBRTC_CALL_H_