mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 00:57:49 +01:00
c0d44d9d63
The implementation here has a number of changes that force the callers that called the "channel" functions into specific interfaces rather than just letting C++ take care of it; this should go away once there stops being a common implementation class for those interfaces. Bug: webrtc:13931 Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38888}
105 lines
4 KiB
C++
105 lines
4 KiB
C++
/*
|
|
* 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 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;
|
|
|
|
// Temporary fix while MediaChannel is being reconstructed
|
|
virtual MediaChannel* media_channel() const = 0;
|
|
virtual MediaSendChannelInterface* media_send_channel() const = 0;
|
|
// Typecasts of media_channel(). Will cause an exception if the
|
|
// channel is of the wrong type.
|
|
virtual VideoMediaSendChannelInterface* video_media_send_channel() const = 0;
|
|
virtual VoiceMediaSendChannelInterface* voice_media_send_channel() const = 0;
|
|
virtual MediaReceiveChannelInterface* media_receive_channel() const = 0;
|
|
// Typecasts of media_channel(). Will cause an exception if the
|
|
// channel is of the wrong type.
|
|
virtual VideoMediaReceiveChannelInterface* video_media_receive_channel()
|
|
const = 0;
|
|
virtual VoiceMediaReceiveChannelInterface* voice_media_receive_channel()
|
|
const = 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_
|