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/*
* Copyright 2018 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 PC_CHANNEL_INTERFACE_H_
#define PC_CHANNEL_INTERFACE_H_
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "media/base/media_channel.h"
#include "pc/rtp_transport_internal.h"
namespace webrtc {
class Call;
class VideoBitrateAllocatorFactory;
} // namespace webrtc
namespace cricket {
class MediaChannel;
class VoiceChannel;
class VideoChannel;
class MediaContentDescription;
struct MediaConfig;
// A Channel is a construct that groups media streams of the same type
// (audio or video), both outgoing and incoming.
// When the PeerConnection API is used, a Channel corresponds one to one
// to an RtpTransceiver.
// When Unified Plan is used, there can only be at most one outgoing and
// one incoming stream. With Plan B, there can be more than one.
// ChannelInterface contains methods common to voice and video channels.
// As more methods are added to BaseChannel, they should be included in the
// interface as well.
// TODO(bugs.webrtc.org/13931): Merge this class into RtpTransceiver.
class ChannelInterface {
public:
virtual ~ChannelInterface() = default;
virtual cricket::MediaType media_type() const = 0;
virtual VideoChannel* AsVideoChannel() = 0;
virtual VoiceChannel* AsVoiceChannel() = 0;
virtual MediaSendChannelInterface* media_send_channel() = 0;
// Typecasts of media_channel(). Will cause an exception if the
// channel is of the wrong type.
virtual VideoMediaSendChannelInterface* video_media_send_channel() = 0;
virtual VoiceMediaSendChannelInterface* voice_media_send_channel() = 0;
virtual MediaReceiveChannelInterface* media_receive_channel() = 0;
// Typecasts of media_channel(). Will cause an exception if the
// channel is of the wrong type.
virtual VideoMediaReceiveChannelInterface* video_media_receive_channel() = 0;
virtual VoiceMediaReceiveChannelInterface* voice_media_receive_channel() = 0;
// Returns a string view for the transport name. Fetching the transport name
// must be done on the network thread only and note that the lifetime of
// the returned object should be assumed to only be the calling scope.
// TODO(deadbeef): This is redundant; remove this.
virtual absl::string_view transport_name() const = 0;
// TODO(tommi): Change return type to string_view.
virtual const std::string& mid() const = 0;
// Enables or disables this channel
virtual void Enable(bool enable) = 0;
// Used for latency measurements.
virtual void SetFirstPacketReceivedCallback(
std::function<void()> callback) = 0;
// Channel control
virtual bool SetLocalContent(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc) = 0;
virtual bool SetRemoteContent(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc) = 0;
virtual bool SetPayloadTypeDemuxingEnabled(bool enabled) = 0;
// Access to the local and remote streams that were set on the channel.
virtual const std::vector<StreamParams>& local_streams() const = 0;
virtual const std::vector<StreamParams>& remote_streams() const = 0;
// Set an RTP level transport.
// Some examples:
// * An RtpTransport without encryption.
// * An SrtpTransport for SDES.
// * A DtlsSrtpTransport for DTLS-SRTP.
virtual bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) = 0;
};
} // namespace cricket
#endif // PC_CHANNEL_INTERFACE_H_