mirror of
https://github.com/mollyim/webrtc.git
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d43af9172b
This CL takes away all usages except for Android code. Low-Coverage-Reason: Refactoring old code Bug: webrtc:15410 Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40554}
195 lines
6.9 KiB
C++
195 lines
6.9 KiB
C++
/*
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* Copyright 2022 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 MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_
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#define MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <functional>
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/functional/any_invocable.h"
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#include "absl/strings/string_view.h"
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#include "absl/types/optional.h"
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#include "api/audio_options.h"
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#include "api/call/audio_sink.h"
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#include "api/call/transport.h"
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#include "api/crypto/frame_decryptor_interface.h"
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#include "api/crypto/frame_encryptor_interface.h"
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#include "api/frame_transformer_interface.h"
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#include "api/media_types.h"
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#include "api/rtc_error.h"
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#include "api/rtp_headers.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_sender_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/sequence_checker.h"
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#include "api/task_queue/pending_task_safety_flag.h"
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#include "api/task_queue/task_queue_base.h"
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#include "api/transport/rtp/rtp_source.h"
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#include "api/video/recordable_encoded_frame.h"
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#include "api/video/video_frame.h"
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#include "api/video/video_sink_interface.h"
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#include "api/video/video_source_interface.h"
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#include "api/video_codecs/video_encoder_factory.h"
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#include "media/base/codec.h"
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#include "media/base/media_channel.h"
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#include "media/base/stream_params.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/dscp.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/socket.h"
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#include "rtc_base/thread_annotations.h"
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// This file contains the base classes for classes that implement
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// the channel interfaces.
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// These implementation classes used to be the exposed interface names,
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// but this is in the process of being changed.
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namespace cricket {
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// The `MediaChannelUtil` class provides functionality that is used by
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// multiple MediaChannel-like objects, of both sending and receiving
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// types.
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class MediaChannelUtil {
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public:
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MediaChannelUtil(webrtc::TaskQueueBase* network_thread,
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bool enable_dscp = false);
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virtual ~MediaChannelUtil();
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// Returns the absolute sendtime extension id value from media channel.
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virtual int GetRtpSendTimeExtnId() const;
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webrtc::Transport* transport() { return &transport_; }
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// Base methods to send packet using MediaChannelNetworkInterface.
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// These methods are used by some tests only.
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bool SendPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options);
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bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options);
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int SetOption(MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option);
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// Functions that form part of one or more interface classes.
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// Not marked override, since this class does not inherit from the
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// interfaces.
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// Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
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// Set to true if it's allowed to mix one- and two-byte RTP header extensions
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// in the same stream. The setter and getter must only be called from
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// worker_thread.
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void SetExtmapAllowMixed(bool extmap_allow_mixed);
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bool ExtmapAllowMixed() const;
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void SetInterface(MediaChannelNetworkInterface* iface);
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// Returns `true` if a non-null MediaChannelNetworkInterface pointer is held.
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// Must be called on the network thread.
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bool HasNetworkInterface() const;
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void SetFrameEncryptor(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
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void SetFrameDecryptor(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor);
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void SetEncoderToPacketizerFrameTransformer(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
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void SetDepacketizerToDecoderFrameTransformer(
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uint32_t ssrc,
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
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protected:
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bool DscpEnabled() const;
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void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
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private:
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// Implementation of the webrtc::Transport interface required
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// by Call().
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class TransportForMediaChannels : public webrtc::Transport {
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public:
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TransportForMediaChannels(webrtc::TaskQueueBase* network_thread,
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bool enable_dscp);
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virtual ~TransportForMediaChannels();
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// Implementation of webrtc::Transport
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bool SendRtp(rtc::ArrayView<const uint8_t> packet,
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const webrtc::PacketOptions& options) override;
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bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
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// Not implementation of webrtc::Transport
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void SetInterface(MediaChannelNetworkInterface* iface);
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int SetOption(MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option);
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bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
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bool rtcp,
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const rtc::PacketOptions& options);
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bool HasNetworkInterface() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return network_interface_ != nullptr;
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}
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bool DscpEnabled() const { return enable_dscp_; }
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void SetPreferredDscp(rtc::DiffServCodePoint new_dscp);
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private:
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// This is the DSCP value used for both RTP and RTCP channels if DSCP is
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// enabled. It can be changed at any time via `SetPreferredDscp`.
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rtc::DiffServCodePoint PreferredDscp() const {
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RTC_DCHECK_RUN_ON(network_thread_);
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return preferred_dscp_;
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}
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// Apply the preferred DSCP setting to the underlying network interface RTP
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// and RTCP channels. If DSCP is disabled, then apply the default DSCP
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// value.
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void UpdateDscp() RTC_RUN_ON(network_thread_);
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int SetOptionLocked(MediaChannelNetworkInterface::SocketType type,
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rtc::Socket::Option opt,
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int option) RTC_RUN_ON(network_thread_);
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const rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> network_safety_
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RTC_PT_GUARDED_BY(network_thread_);
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webrtc::TaskQueueBase* const network_thread_;
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const bool enable_dscp_;
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MediaChannelNetworkInterface* network_interface_
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RTC_GUARDED_BY(network_thread_) = nullptr;
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rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) =
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rtc::DSCP_DEFAULT;
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};
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bool extmap_allow_mixed_ = false;
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TransportForMediaChannels transport_;
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};
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} // namespace cricket
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#endif // MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_
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