99c8a80b8e
This changes the notification to a single std::function pointer instead of being a sigslot::signal1<> collection. Summary: * Remove SignalFirstPacketReceived_, the last sigslot member variable. (still inherits from sigslot::has_slots<>) * BaseChannel doesn't post to the signaling thread anymore. The only reason that remains for the signaling_thread_ variable, is for thread checking. * Remove BaseChannel's reliance on MessageHandlerAutoCleanup (still inherits from MessageHandler) RtpTransceiver is the consumer of this event. That class is also the class that sits between the PC classes and the channel object, holding a pointer to the channel and managing calls that come in on the signaling thread, such as SetChannel. The responsibility of delivering the first packet received on the signaling thread is now with RtpTransceiver: * RtpTransceiver always requires a ChannelManager instance. Previously this variable was sometimes set, but it's now required. * Updated tests in rtp_transceiver_unittest.cc to include a ChannelManager as well as fix them to include call expectations for mock sender and receivers. Bug: webrtc:11993, webrtc:11988 Change-Id: If49d6be157cd7599fa6fe3a42cd0a363464e3a74 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215979 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33853}
77 lines
2.5 KiB
C++
77 lines
2.5 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 "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 cricket {
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class MediaContentDescription;
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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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// TODO(deadbeef): This is redundant; remove this.
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virtual const std::string& transport_name() const = 0;
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virtual const std::string& content_name() 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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// Returns the last negotiated header extensions.
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virtual RtpHeaderExtensions GetNegotiatedRtpHeaderExtensions() const = 0;
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protected:
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virtual ~ChannelInterface() = default;
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};
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} // namespace cricket
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#endif // PC_CHANNEL_INTERFACE_H_
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