2f55370634
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808. Reason for revert: Added a field trial that allows landing the code without affecting performance in prod. This CL also incorporates subsequent CLs that also had to be reverted. Original change's description: > Revert "Use two MediaChannels for 2 directions." > > This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f. > > Reason for revert: Quality regression detected. > > Original change's description: > > Use two MediaChannels for 2 directions. > > > > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other. > > > > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it. > > > > Bug: webrtc:13931 > > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39340} > > No-Try: true > Bug: webrtc:13931 > Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39445} Bug: webrtc:13931 Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39494}
107 lines
3.9 KiB
C++
107 lines
3.9 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 MediaChannel;
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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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// 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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