36fafc8827
This allows callers to differentiate on whether they need the channel for sending or receiving purposes. Note: This CL is incomplete, in that many places cast the pointers to the concrete subclasses "VideoMediaChannel" and "AudioMediaChannel", which are not split into sending and receiving APIs. The long term goal is to make two MediaChannel-like class APIs, with distinct implementations, and let the RtpSender and RtpReceiver manage those objects, rather than keeping them in the RtpTransceiver. Bug: webrtc:13931 Change-Id: I8d56defe2287bd6552b71571cc6a5ec842927fa4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287040 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38844}
101 lines
3.8 KiB
C++
101 lines
3.8 KiB
C++
/*
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* Copyright 2018 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 PC_CHANNEL_INTERFACE_H_
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#define PC_CHANNEL_INTERFACE_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/strings/string_view.h"
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#include "api/jsep.h"
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#include "api/media_types.h"
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#include "media/base/media_channel.h"
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#include "pc/rtp_transport_internal.h"
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namespace webrtc {
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class Call;
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class VideoBitrateAllocatorFactory;
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} // namespace webrtc
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namespace cricket {
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class MediaContentDescription;
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struct MediaConfig;
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// A Channel is a construct that groups media streams of the same type
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// (audio or video), both outgoing and incoming.
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// When the PeerConnection API is used, a Channel corresponds one to one
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// to an RtpTransceiver.
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// When Unified Plan is used, there can only be at most one outgoing and
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// one incoming stream. With Plan B, there can be more than one.
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// ChannelInterface contains methods common to voice and video channels.
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// As more methods are added to BaseChannel, they should be included in the
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// interface as well.
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// TODO(bugs.webrtc.org/13931): Merge this class into RtpTransceiver.
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class ChannelInterface {
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public:
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virtual ~ChannelInterface() = default;
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virtual cricket::MediaType media_type() const = 0;
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virtual MediaSendChannelInterface* media_send_channel() const = 0;
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// Typecasts of media_channel(). Will cause an exception if the
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// channel is of the wrong type.
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virtual VideoMediaChannel* video_media_send_channel() const = 0;
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virtual VoiceMediaChannel* voice_media_send_channel() const = 0;
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virtual MediaReceiveChannelInterface* media_receive_channel() const = 0;
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// Typecasts of media_channel(). Will cause an exception if the
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// channel is of the wrong type.
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virtual VideoMediaChannel* video_media_receive_channel() const = 0;
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virtual VoiceMediaChannel* voice_media_receive_channel() const = 0;
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// Returns a string view for the transport name. Fetching the transport name
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// must be done on the network thread only and note that the lifetime of
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// the returned object should be assumed to only be the calling scope.
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// TODO(deadbeef): This is redundant; remove this.
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virtual absl::string_view transport_name() const = 0;
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// TODO(tommi): Change return type to string_view.
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virtual const std::string& mid() const = 0;
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// Enables or disables this channel
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virtual void Enable(bool enable) = 0;
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// Used for latency measurements.
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virtual void SetFirstPacketReceivedCallback(
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std::function<void()> callback) = 0;
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// Channel control
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virtual bool SetLocalContent(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string& error_desc) = 0;
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virtual bool SetRemoteContent(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string& error_desc) = 0;
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virtual bool SetPayloadTypeDemuxingEnabled(bool enabled) = 0;
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// Access to the local and remote streams that were set on the channel.
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virtual const std::vector<StreamParams>& local_streams() const = 0;
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virtual const std::vector<StreamParams>& remote_streams() const = 0;
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// Set an RTP level transport.
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// Some examples:
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// * An RtpTransport without encryption.
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// * An SrtpTransport for SDES.
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// * A DtlsSrtpTransport for DTLS-SRTP.
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virtual bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) = 0;
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};
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} // namespace cricket
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#endif // PC_CHANNEL_INTERFACE_H_
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