diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/pc/channel_interface.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/channel_interface.h')
-rw-r--r-- | third_party/libwebrtc/pc/channel_interface.h | 108 |
1 files changed, 108 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/channel_interface.h b/third_party/libwebrtc/pc/channel_interface.h new file mode 100644 index 0000000000..783755e6fe --- /dev/null +++ b/third_party/libwebrtc/pc/channel_interface.h @@ -0,0 +1,108 @@ +/* + * 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 VoiceChannel; +class VideoChannel; +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; + + virtual VideoChannel* AsVideoChannel() = 0; + virtual VoiceChannel* AsVoiceChannel() = 0; + + // Temporary fix while MediaChannel is being reconstructed + virtual MediaChannel* media_channel() = 0; + virtual MediaSendChannelInterface* media_send_channel() = 0; + // Typecasts of media_channel(). Will cause an exception if the + // channel is of the wrong type. + virtual VideoMediaSendChannelInterface* video_media_send_channel() = 0; + virtual VoiceMediaSendChannelInterface* voice_media_send_channel() = 0; + virtual MediaReceiveChannelInterface* media_receive_channel() = 0; + // Typecasts of media_channel(). Will cause an exception if the + // channel is of the wrong type. + virtual VideoMediaReceiveChannelInterface* video_media_receive_channel() = 0; + virtual VoiceMediaReceiveChannelInterface* voice_media_receive_channel() = 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_ |