blob: 77cd3d2690103274a5c01854e6ab93dc9f5c27d9 [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 CALL_CALL_H_
#define CALL_CALL_H_
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "api/media_types.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
namespace webrtc {
// 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 {
using Config = CallConfig;
struct Stats {
std::string ToString(int64_t time_ms) const;
int send_bandwidth_bps = 0; // Estimated available send bandwidth.
int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
int64_t pacer_delay_ms = 0;
int64_t rtt_ms = -1;
static Call* Create(const Call::Config& config);
static Call* Create(const Call::Config& config,
Clock* clock,
std::unique_ptr<ProcessThread> call_thread,
std::unique_ptr<ProcessThread> pacer_thread);
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(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config) = 0;
virtual VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller);
virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
virtual VideoReceiveStream* CreateVideoReceiveStream(
VideoReceiveStream::Config configuration) = 0;
virtual void DestroyVideoReceiveStream(
VideoReceiveStream* receive_stream) = 0;
// In order for a created VideoReceiveStream to be aware that it is
// protected by a FlexfecReceiveStream, the latter should be created before
// the former.
virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) = 0;
virtual void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* 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;
// This is used to access the transport controller send instance owned by
// Call. The send transport controller is currently owned by Call for legacy
// reasons. (for instance variants of call tests are built on this assumtion)
// TODO(srte): Move ownership of transport controller send out of Call and
// remove this method interface.
virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0;
// Returns the call statistics, such as estimated send and receive bandwidth,
// pacing delay, etc.
virtual Stats GetStats() const = 0;
// TODO(skvlad): When the unbundled case with multiple streams for the same
// media type going over different networks is supported, track the state
// for each stream separately. Right now it's global per media type.
virtual void SignalChannelNetworkState(MediaType media,
NetworkState state) = 0;
virtual void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
virtual void SetClientBitratePreferences(
const BitrateSettings& preferences) = 0;
virtual ~Call() {}
} // namespace webrtc
#endif // CALL_CALL_H_