68f4e27794
It works in the same way as the first packet received callback and can be used for latency measurements. One important detail is that RTCP and probe packets are excluded from triggering the callback. Bug: b/375148360 Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43309}
108 lines
4.0 KiB
C++
108 lines
4.0 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 <functional>
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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 VoiceChannel;
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class VideoChannel;
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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 VideoChannel* AsVideoChannel() = 0;
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virtual VoiceChannel* AsVoiceChannel() = 0;
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virtual MediaSendChannelInterface* media_send_channel() = 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 VideoMediaSendChannelInterface* video_media_send_channel() = 0;
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virtual VoiceMediaSendChannelInterface* voice_media_send_channel() = 0;
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virtual MediaReceiveChannelInterface* media_receive_channel() = 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 VideoMediaReceiveChannelInterface* video_media_receive_channel() = 0;
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virtual VoiceMediaReceiveChannelInterface* voice_media_receive_channel() = 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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virtual void SetFirstPacketSentCallback(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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