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The interface is implemented by the ChannelManager and contains methods to create and destroy media channel objects as used by a transceiver. This will subsequently allow us to delete the channel objects from the transceiver class where ownership really lies rather than from the outside - which is currently required by some tests that keep channel objects on the stack. We'll furthermore be able to do the destruction asynchronously without additional Invoke()s as we do now which will remove an Invoke when making sdp changes. With introducing the interface, the following simplifications were made: * ChannelManager constructed on the signaling thread. Before, there was an Invoke in the context class, which existed for the purposes of calling MediaEngine::Init() (which in turn is only needed for the VoiceEngine). This Invoke has now been moved into the CM (more tbd). * The CM now has a pointer to the signaling thread (since that's the construction thread). That allows us to remove the signaling thread parameter from the CreateFooChannel methods. * The ssrc_generator (UniqueRandomIdGenerator) instance for SSRCs moved from SdpOfferAnswerHandler to the CM, as it's always used in combination with the CM. This simplifies the CreateFooChannel methods as well as a couple of other classes that have a CM dependency. * Removed DestroyFooChannel related code from SdpOfferAnswerHandler since the channel type detail can be taken care of by the CM. Bug: webrtc:11992, webrtc:13540 Change-Id: I04938a803734de8489ba31e6212d9eaecc244126 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247904 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35766}
113 lines
3.6 KiB
C++
113 lines
3.6 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 <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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class VideoChannel;
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class VoiceChannel;
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struct MediaConfig;
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// ChannelInterface contains methods common to voice, video and data 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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class ChannelInterface {
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public:
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virtual cricket::MediaType media_type() const = 0;
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virtual MediaChannel* media_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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protected:
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virtual ~ChannelInterface() = default;
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};
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class ChannelFactoryInterface {
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public:
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virtual VideoChannel* CreateVideoChannel(
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webrtc::Call* call,
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const MediaConfig& media_config,
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const std::string& mid,
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bool srtp_required,
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const webrtc::CryptoOptions& crypto_options,
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const VideoOptions& options,
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webrtc::VideoBitrateAllocatorFactory*
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video_bitrate_allocator_factory) = 0;
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virtual VoiceChannel* CreateVoiceChannel(
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webrtc::Call* call,
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const MediaConfig& media_config,
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const std::string& mid,
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bool srtp_required,
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const webrtc::CryptoOptions& crypto_options,
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const AudioOptions& options) = 0;
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virtual void DestroyChannel(ChannelInterface* channel) = 0;
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protected:
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virtual ~ChannelFactoryInterface() = default;
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};
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} // namespace cricket
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#endif // PC_CHANNEL_INTERFACE_H_
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